Finished tracking the final song!

This week in labs we wrapped up the tracking for out third and final song of the semester. Vocals and guitars were the goal this week. Amber came back down to Monterey to record to the second section of our song, which is the only place we decided to have vocals partly because the music is more technical throughout, and is more of an instrumental tune to begin with. The previous night was spent doing some preproduction work in chunking out melodies and lyrics for the song. We used an AKG 414 that we had set up in room 185, and sent the signal through the Neve preamp. Some ideas for vocal harmonies were brought, and we came up with some decent on the spot harmonies that some of us seemed to be hearing in our heads. We tracked some acoustic guitar next, for the third section of the song. This whole section is 5/4, and some people aren’t as comfortable playing in this time signature, so practice and preparation was necessary. The strings on the guitar probably needed changing, because we had some problems staying in tune, and had to re-tune every other take. After trying to get a decent take down, we decided to call it a day and shoot for it again on Thursday. Thursday morning went a lot better and we got some good takes down fairly quickly. We used the rest of out lab time to try some mixing out of the box, which is what we have been doing for the past week or so now.
For mixing out of the box, we bypass 2track and come right out of the board. We assigned specific channel outputs in protols corresponding to the patchbay. Using the patch bay, go out of PT into line 1 inputs on the channels closest to the center of the console (23, 24, 25). Make sure the mix bus and Line 1 buttons on the channel strip are on, and the aux master mix bus buttons and 2trk are off. Fader at unity gain. To send an outboard reverb to a channel, patch out auxsends into the effects processor, and out the effect into echo returns. Echo faders 1 2 at unity gain, panned hard left and right. Push in the prefader button in the aux section of the board. Be sure that you’ve turned up the aux1 knob on the instrument channel strip. The channel being mixed on is set to prefader and unity gain. For more editing tips in protocols, trim all the tails and empty space in all regions, for a quick mix, make sure automation is off and you have a basic stereo of the song.

Mixing out of the Box

With a new headphones matrix set up in the studio, we can now have individual mixes for the recording artists. We can customize these mixes for each player as they track. This is very beneficial to the recording process, and creates a comfortable aural enviroment. Mixing out of the box is a really awesome concept and I am finally glad to say that I am gaining an understanding of signal flow in complex ways that I didn’t know before. Instead of using the mouse to move faders and dial in digital effects, we are using our hands on the faders and physically turning knobs for sound quality purposes, mixing out of the box. Mixing “in the box” with all digital effects and mouse clicking really takes away from the aural experience. There is much visual stimulation happening, and can alter the way a person decides where to begin with a mix. Down to the meat… Go into your protools session and assign specific channel outputs. Using the patch bay, go out of PT into line 1 inputs on the channels closest to you on the board (34, 35, 36). Make sure you are on the mix bus on the channels strip now, and off the aux master mix bus. Line 1 inputs on the channel strip activated, and fader at unity gain. To send an outboard reverb to a channel, patch out auxsends into the effects processor, and out the effect into echo returns. Echo faders 1 2 at unity gain, panned hard left and right. Push in the prefader button in the aux section of the board. Be sure that you’ve turned up the aux1 knob on the instrument channel strip. The channel being mixed on is set to prefader and unity gain. In labs this week we recorded the French horn and both trumpet parts. The recording went smoothly but the player forgot the music so I had to sit there with him and teach it to him using the piano. It took up some time but luckily the takes went well. We used a royer 121 in room 185 in the vocal booth, to get a dry sound. This has synthesizers in it, so I wanted to have some really cool effects on dry tracks. On Thursday we got recordings of the guitar layers done for the first section of the songs. We used the 121 once again since we like it so much. Practicing during the week really helped the recording go a lot smoother. Charting out the song is always a good idea and has only benefited the groups tracking sessions. I am pleased with the way things are going in the group.

Compression! and more.... RECORDING!

This week, there was no class Monday, so on Wednesday we covered a lot about compression. Compressors are used to smooth out the dynamic range of a performance, typically on instruments with a wide dynamic range like vocals, drums, and bass. These dynamic range processors can boost/attenuate volume indifferences within an audio signal. The benefits of using compression can include a smoother sounding and more consistent performance, which is much easier to listen to that something with too much dynamic range. It can make soft parts louder and loud parts softer. It changes the roominess of a sound, and changes the tone of the source. A compressor has a few parameters: The threshold is a user-defined setting measured in dB of how much audio is being affected. If there is a threshold of -15dB, any audio that is sounding above that threshold will be compressed, and the audio underneath it will remain unaffected by the compressor. Ratio is another aspect of a compressor. It defines the comparison of unaffected audio to compressed audio. A 3:1 ratio means that for every 3dB of audio over the set threshold, it will be heard at a volume of 1dB over the threshold. Any ratio over 10:1 is considered the use of a limiter. Attack time (in milliseconds) is the time it takes a compressor to react to the audio signal above the threshold. This setting can make the difference between a crack or thuck sound in a snare drum. A fast attack time creates a soft, pillowy drum sound. The release time is the time it takes a compressor to return the signal back to normal, or unaffected. Setting a longer release can bring up the quieter parts in the tail of an audio signal. The output of the compressor controls the audio level after compression.
This week in labs we did some more recording for the third song that I have composed for the group! Tuesday morning was recording drums for the first section of the song. We set up the usual, D112 on the kick, e22’s on snare top and bottom, MD 421’s for toms, and the 414’s as overheads. Tracking these drums was fun for me, because I wrote this song in mostly odd time signatures and the grooves turned out, well, pretty groovy! This took the whole lab because I wanted to make sure we had one of the best takes possible for that day, and we accomplished that. On Thursday morning, I brought in my MIDI controller again to track some more parts for the sections of the song. We hooked the keyboard up to a laptop that has Logic as the DAW, because I had found this really cool patch that I wanted to use for this song. The only thing is this patch has a delay, phaser, and I EQ’d it a little bit. This just means that I had to play tight, and with no punching in really! With fixed effects, there is no going in and removing them after it has been recorded, so it needed to be tight. With what little time I had before we were supposed to track, I stayed up late the previous night practicing the parts, and in turn the tracking went a lot smoother than I know it would have had I not sat down and worked on it. We got the first section done last week so we had to record the transition into the second section, the transition into the third section, and the entire third section. We set everything up the same way as last time, running out of the laptop into an interface, and then out of the interface into a guitar amp, and also into a R and L direct in. We used the 414 and the Royer 121 again, even though we knew we liked the way the Royer sounded by itself. After getting all that tracking done, we got the other room set up to record a real piano for the middle section of the song. We miked it with two 414’s, and it took one or two takes for me to nail the piano part. Onto the viola with what little time we had left! We quickly moved a 414 from the piano and set up a new audio track for viola. Taylor checked out sheet music I had created for him a few minutes before, and nailed the part in a couple of takes. That concluded the day.

EQ and more recording!!

This week EQ was the focus. Equalization of frequencies is very important in the aesthetics of sound, and it is a tool that has a very fine line of too much or too little. Equalizers modify the amplitude of a selected part of the frequency spectrum of an audio signal. It is used to correct audio deficiencies, and is also used in creative ways for effects. Flat frequency response refers to when a microphone receives a signal with the same sensitivity for all frequencies. EQ’s have different filter types. They isolate and boost or cut certain frequencies and harmonics in various ways. A low pass filter os one that filters out all the high frequencies, and lets the lows pass through. You can set the cut off point at a specific point, and it rolls off the frequencies at a certain amount of decibels per octave, depending on the setting. A high pass filter does the opposite of a low pass filter. It filters out all the low end and lets the highs pass through. Low pass filters are good to use on bass guitars or low end instruments, because high frequencies aren’t typically present, or aren’t needed. A band pass filter is one that grabs a specific band of frequencies, and cuts or boosts them. Widening the Q or bandwidth will allow more frequencies through the filter, while narrowing it hones in on a specific frequency. The degree of boost or cut is call the resonance of the filter and is expressed in dB. Band pass filters are fully sweepable. George Massenburg was the inventor of the fully parametric EQ. This is a notch filter. It is great to use when there is a hum in a track, or a certain frequency or overtone keeps popping out in a mix. Proper EQ starts with the person or artist tracking. Hopefully they know something about EQ and tone, otherwise its up to the engineer to work the magic. The artist, their attitude that day, the instruments used, proper working cables, mic choice and placement, and the mic pre are all variables when it comes to getting a fitting EQ before tracking. EQ’s have 3 parameters: Frequency, gain, and Q or bandwidth. The frequency parameter enables the selection of a specific frequency to boost or cut. Gain is a measurement in dB of the overall EQ. The bandwidth adjusts which frequencies are allowed through the filter or not. When using EQ, it is usually better to use subtractive EQ rather than additive EQ. Adding EQ causes a digital plug-in to create extra sounds that weren’t actually in the original recording. To get a beefier low end, pull down the highs rather than boosting the lows! Complementary EQ is another type. This is were cutting frequency holes in certain instruments is crucial to allow the audibility of other fundamentals, harmonics, and overtones. If two guitars are dominating the mix, and the lo mids are too high, slap an EQ on one of the guitars and bring down the lo mids. If you have several guitars going, you don’t need the fullness of all of them, just one or two of them. Cut the interfering frequencies out of the others. In order to find where holes need to be drawn, set a notch EQ and exaggerate the gain. Sweep the frequency spectrum to find the stand out frequency that you don’t want. Pull the gain into negative dB and the obnoxious ringing should minimize significantly, or hopefully disappear. Before adding EQ, listen to the instrument in context with all the other instruments. Soloing the track won’t give you a good reference to what you are EQ’ing. You may spend quite some time doing EQ on a track solo’ed and make it sound really good, only to find out that when you bring the rest of the mix back in, it doesn’t fit and ends up masking other frequencies. EQ really should be a last resort thing, as it is always good to be aware of all of the variables of recording and to get a good recording the first time.
This week in labs we finished up the first tune that we needed to re-track. We tracked vocals on Tuesday morning in room 185 this time, using the wonderful AKG 414. Things went pretty smoothly and the song is coming around! It didn’t take very long to get the vocals down, so we spent the rest of the lab time mixing the song. We cleaned up the start and end of the tracks, and set a basic pan to get a temporary stereo mix. We adjusted levels so everything was relative in volume. On Wednesday evening we recorded drums for the new song. We did the standard micing that we know and love. The D112 on the kick, e22’s on the top and bottom snare, MD421’s on the toms, and the 414’s for the overheads. The first section of this song is in 7/4, and was interesting to record. The next morning we recorded keyboards, and we ran my midi controller through a cool synth patch in logic on a laptop. We went out of the laptop, through an interface, and out the interface into the millennia for a direct in signal. We also went out the interface into a guitar amp that was miked with a 414 and a Royer 121. The 121 ended up sounding better and warmer than the 414. This was my first experience recording midi as audio, and I was super stoked about the whole process, and I cant wait to record more!

The week after Spring break!

This week was an interesting one. It was an eye-opener for a lot of people, and an overall learning experience for everyone in the group. Monday was the day we got our midterms back. I had a feeling that I did well on it, I went in feeling comfortable with the information I knew. My grade turned out to be 96%. This was really nice to see! After reviewing the test and discussing the answers, we went into the control room to listen to each of the group’s mixes for the third recording assignment. The tracking process went well, and we hadn’t done much mixing because time was up and we had a more than decent sounding tracking session as it was. Apparently I was unaware that at the end of our final lab before the song was due, one of the group members spent the rest of the lab time doing some mixing while the others were cleaning up the studio and equipment. During the class listening, this new mix was pretty crappy. A guitar with a SansAmp digital plug-in was used for distortion, and a bad tone was chosen. Somehow, there was a fade on the bass track drawn in from start to finish, and by the time the end of the song came, the bass was out. A “too wet” reverb was put on the vocals, and the viola was mixed way too high. Had we all been in the studio when this was happening, I’m sure it would have gone differently, and slightly better.
Since the majority of the class didn’t do well on the test, our next lab we were required to physically review signal flow and routing a pre fade reverb to a vocalist’s headphones mix, so as to send the vocalist reverb and not hear the effect in the control room monitors. I was confident in knowing how to do it, and that I could do it, but I didn’t know that could do it rather quickly. I realized that I was having an easier time on the board, so I only practiced one time and allowed the others in the group to get more time that day. The process involved hooking up a headphones monitoring system in the MIDI lab, setting up an SM58 into channel 1. Different people went and tested the mic as a vocalist as we were routing everything, making sure they were getting signal, reverb and talkback.
Wednesday night was an extra night that we had booked to re-track the second recording assignment. We were starting from the ground up with tracking the instruments in the song, but it was a very productive night. We started out with drums, using a D112 on the kick inside, top and bottom snare drum with two e22’s. We used two 414’s with a cardioid setting for the overheads, and two MD421’s on the toms. Chris got the drums tracked in a couple takes. The next instrument was bass which we had done three takes of. We had hired Nick Reves this time, and he did an excellent job, especially for just reading off of the chart we had handed him 15 minutes before. We decided to go direct into the Millenia as we had done before. Next up was the acoustic guitar, which we used a 414 as well. We recorded that in the big room, and it came out sounding nice. We were then on to tracking piano, all in the same night. The vibe had been great the whole time, and we were really cranking stuff out. We used the 414’s on the piano, and only a few takes were needed to nail the piano parts.
The next morning we recorded electric guitar with a royer 121 back in the MIDI lab, with sonnex surrounding the guitar cab. We tracked and decided to make a few quick arrangement decisions for the better of the song, and everyone turned out liking it. Next we moved onto viola, the last instrument that needed to be tracked, besides vocals. Taylor managed to nail it in a couple takes, and we were finished tracking the music! In two days! I was super stoked because previously it had taken 3 weeks to get done what we did so quickly. Lately there has been some turmoil in our group, and clashing personalities and issues. Being late, and taking control of a situation were two main issues that needed to be addressed among the group. We got together with the group, while Drew facilitated a meeting to talk about our interpersonal issues that had come up, and other issues regarding progress in the group. Everyone had seemed to come to an understanding of what had been going on, and that it had to stop for the sake of the group. These next few weeks will hopefully show some group dynamic changes for us.

Crosstalk, bleed, out of the box!

Today’s lesson was on crosstalk, and how to prevent it. My group has been tracking drums, bass, piano, guitars, and vocals all over again, for the 4th time because we had click track bleed that was recorded onto more than half of the drum tracks. Just for a simple demonstration, we put 1 overhead AKG 414 cardioid over the kit. We set up a headphones monitoring pathway. We set up a click track and recorded a little bit of drums. Listening back, there was no bleed. When monitoring in the control room, the click track should be very low in volume, almost inaudible. My group had apparently been tracking with the click at a volume loud enough that two signals running next to each other had crosstalk.
Today we also learned how to route outboard processors through the board, and record a master mix running the audio through the board. The initial audio from a track or audio file runs out of the computer through its core and excel cards. It then travels into the interface (192’s) with AD/DA conversions happening. The interface is directly routed to the patch bay. Group all the drums together and send them to a stereo aux track labeled drums (input bus 3-4). Send signal out A 1-2 into line inputs 39 and 40. Route 2 more cables from channel insert sends 38-40 to Distressor 1-2, and then route out of the Distressor into the channel insert returns 39 and 40 Route the kick drum from interface’s output (A-1) into line inputs 1 on any channel, in this case, we are using channel 39 and 40. On channel strips 39 and 40 on the board, make sure faders are at unity gain, center pan, and the line 1 and mix bus buttons are pushed in. Create a stereo audio Master track. Input is B 1-2, Output is A 7-8 because A 1-2 is being used to monitor. Adjust the amount of distressor using the aux knobs on the master output and channels 39 and 40, and on the Distressor itself.

The amazing Tom Dowd

In class on wednesday, we watched a documentary "Tom Dowd and the Language of Music". I have not heard of this man before today, but I know of and own many things that he has contributed to or helped create. Tom was growing up and going to school in New York. His mother was a singer, and his father staged big shows and theatres like the Roxy and the Capitol. Among being the member of the school band, he was highly interested in the study of Physics. After high school, Tom was drafted into the military, and he later on began to work for what was known as the Manhattan Project, the driving force behind the A-bomb that was dropped on Japan. In hopes to get his work for the nuclear industry credited towards school, the denied him and he thought about going back to get his degree. The school was going to make him take a bunch on classes based on the principles of 30's physics, but he was well beyond that, contributing to the discovery and development of 1950's physics. He decided not to continue on with school, and got a summer job at a recording studio, were he found his true passion. Shortly after, Tom started to engineer mixes and work with artists like Aretha Franklin, The Drifters, The Allman brothers, just to name a few. Tom was noticing that studios had many limitations of capturing a good recording. Back then, while recording you had to mix on the fly, or while recording was happening. From the lack of equipment, to the unprofessionalism of the musicianship Tom saw many things that needed changing. Later on in the DVD I would find Tom in the studio with an artist, and he could just listen and know what chord the song needed to go to to sound right. Many artists commented on what an amazing ear he had, not to mention his ability to get a great sound. He would walk around in the studio, having people playing, listening, then he walk back into the control room and tweek something he thought needed it. He later began working with Jerry Wexler of Atlantic studios, and convinced Jerry that they should be recording with 8 track! Dowd was one of the first people ever to mix 4 track down to 2 track, and to begin recording with 8 track. Tom was working with Cream for a while, and continued on his musical and production journey with Eric Clapton. Some of the most important things to Tom for making a good record, was great players and a good vibe. Wether in the studio or backstage at a show, Tom was always pumping up artists to play their best, like a sports coach. If you do not have a good vibe, a good performance will not be captured. Tom was pretty much the most significant person in the music production industry as we know it today. With out him many of the options available to us today would be non-existent.

Tracking Vox and Drums

This week in our labs we spend our first chunk of lab re-tracking vocals with a new singer. As noted in earlier blogs, we had screwed up our production schedule by hiring a singer that wasn't fit for the job. We went ahead and tracked with this singer anyways, and of course, the tracks came out unusable. With a new singer ready to record, we set up the vocal booth in room 100. We put sonnex on the top of the booth as a sort of roof, and behind the singer was a thick blanket creating a third wall. Our vocal set up was triangular shaped, and the singer was going to sing directly into one of the corners of it. We decided to use two mics, just so we'd have the option to choose between which mic we liked best. The AKG 414 was our first choice, followed by the Royer 121. We routed the mics into the Neve preamp. Tracking the vocals ended up going really well, with only a few takes needed to capture what we were looking for. After finishing tracking we still had about 30 minutes of lab time, so we tried a few different harmonies over the melody. After bouncing and saving our session, we backed up or files. Some how the harmony tracks got corrupted, but the song was still great with the main melody.
Our second section of lab time was spent tracking drums for our new piano song. Early in the morning, we set up the kit as we normally do in room 100. We put 2 sides of the vocal booth on either side of the kit, and a sound panel behind the kit. The d112 on the kick, the Josephson e22 on the snare (top and bottom), two MD-421's for the toms, and two AKG 414's for the over heads. We also did mid-side miking a few feet away from the kit with the 4050's. 8 recording tracks for a 4 piece kit! After we got all set up and ready to track, we uploaded a scratch piano track that I had programmed at home on Logic and put it on an audio track that was sent to the headphones mix. Everyone was very helpful in the set up and take down of all the mics, mic stands, cables, patch cables, and the whole deal. It took a few times to get the drums down, and in the end the drummer was still stressing about how it could have been tracked better. Although this song is considered completely written, some are still wanting to add parts to the song that just dont seen to fit with the song. We all still need to discuss this and make sure that we are doing what is best for the song. I'm not trying to shit on anyone's ideas here, I encourage them. However if a part doesn't fit the song, it just doesn't fit the song.
This week in class we started going in depth about channel strips and the gadgets on them. The routing buttons at the top allow sends to the group bus. The mic/line inputs are the next thing down on the channel strip. Here, a gain stage at the mic pre happens. THere is a button for phantom power as well. Right below that is a four band EQ with high, high mids, low mids, and lows, and it has a low pass filter button. There are 8 aux sends, 4 mono and 2 stereo and can be switched between pre and post fader. This allows you to send effects through the board, affecting the signal before or after the fader. The mute button on the channel strip mutes the signal, making it inaudible until it would like to be heard again. A signal peak LED indicator shows when a signal is reaching between +10 to +18 dB.

(Audio Terms) test

Sound waves can be simple or complex. (Propagation) is the production or movement of something. Wave propagation is just a complex way of describing how sound waves move in the directions that they do. (Waveform) is a visual realization of sound waves, generally represented by a graph. These waveforms have different shapes and lengths including an s-shaped wave (the sine wave), square waves, triangle waves, sawtooth waves, rectangle waves, and even more complex shapes. When many simple waves create a complex wave, they are called (partials). More in depth, (overtones) are triggered when a fundamental pitch is sounded. (Harmonics) are other tones that occur when a pitch is heard because their frequencies double, and when they are doubled we hear the octave. (Wavelength) is measured in feet. A (sine wave) is a simple sound wave, generating a monotone. Waves travel at a (velocity) of approximately 1,130 ft/sec. (Wavelength = V/f) is the equation for calculating a waves length in feet. (Frequencies), synonymous with pitch, are measured in cycles per second and are represented as (hertz (Hz)). The (fletcher-munson curve) is a graphical representation of how we hear frequencies and how sensitive we as humans are to certain frequencies. Low end frequencies around 60hz are less audible to us than mid range frequencies around 1,000Hz. A (cycle) is one complete revolution of a sound wave from its positive state to its negative state, then back to 0. (Sound pressure waves) are the movement of sound waves through the air. Vibrations in the air are cause a change in air pressure. The sound pressure waves are measured by a microphone in air or a hydrophone in the water. Sound pressure waves are then transduced into electrical waves by the diaphragm of a microphone. (SPL or sound pressure level is a measurement of sound pressure in (decibels (dB)). Decibels are a measurement of the amplitude of frequencies. On the x – y axis, y represents dB, and x represents frequency. Amplitude is a measurement of how loud a frequency is. Frequencies have (harmonic content), which means that they are sonically attracted to one another. The higher the amplitude, the louder the sound is going to be. According to the laws of pressure, when pressure is doubled it will increase by 6dB. As the speed of sound changes, rarefaction occurs. (Compression) is the physical act of sound waves being squeezed into a certain space. Compression of sound increase as sound travels closer to the ear. (Rarefaction) happens when a sound changes environments, and is the actual bending of sound waves. This will change how sound is perceived by the ear. (Phasing) is when sound waves come in contact with one another. When sound waves cross at the same points, they are in (phase). If the signals cross each other at different points on the 0 line, they are considered to be out of phase, also called (phase shifting). When a signal is out of phase it may start sounding thinner and washy. (Acoustics) are the different characteristic qualities of sound in space. Many variables include but aren’t limited to: (reflections) of sound, which are sound waves that have bounced off of surfaces at various angles, known as (angles or incidence/reflection), and when waveforms come in contact with one another, it can create (standing waves) which are measured as a difference in air pressure, not to be confused with phase which occurs before reflections. (Flutter slap echo) is when a sound bounces off of many walls before reaching the transducer. After standing waves comes (room modes) which are disturbances in low end frequencies. Bass traps in the corner can minimize bouncing off of the back wall. (Acoustic isolation) is removing all the acoustics and sound characteristics of a room by insulating with walls of concrete, and layers of professional sound foam, creating a “dead” room sound. (Vocal booths) are a way of acoustically isolating a singer to capture the closest possible sound, and a dry sound as well, with little acoustics. (Reverb), or reverberation, is the tail end of a sound that you hear in the room after its release (explained later). If you yell in a bathroom, after you stop what you are hearing is reverb. If you do the same thing outdoors the woods, chances are you will not hear much of a reverb because the sound did not have surfaces to bounce off of. (Diffraction) of sound waves happens when the wave travels around a corner or down a hall, or through a doorway. (Diffusers) are devices made of wood or Styrofoam that deflect waves in a mathematical process that will keep them in phase and allow their cycles to complete. Every sound wave has an attack, decay, sustain and release creating a sound (envelope). The (attack) is the start of a transient, or sound, until it peaks in amplitude. The (decay) starts when the signal descends from its peak, and settles into the (sustain) portion of the envelope. This is where the sound wave has a smooth, gradual amplitude decrease. The (release) is the point when the signal comes back to 0dB. (Timbre) is the same as the quality of a sound. How a snare drum sounds versus how a bass or piano sounds are differences in timbre. (Beats) are the organization of sound in real time, on a grid in an aurally pleasing rhythmic fashion. (Frequency balance) is the attenuation and amplification in frequencies, also known as equalization. To attenuate a frequency means dip certain frequencies. An example of (direct sound) would be listening to headphones. Sound is only a couple of inches away from the eardrum, and sound is perceived obviously, directly. (Panning) refers to the stereo image, and hearing audio from the right or left perspective. Pan to the right or left to hear the right side of sound picked up by the microphone. (Near field monitors) are speakers that are placed near the producer, and (far field monitors) are placed significantly farther away. This allows the sound to develop more fully because it has more distance to travel to the ears.

Standard miking techniques, grouping, mixing out of the box, and pre-production!

This week we had a make up class in which we set up 3 different types of overhead mic matrices, ...At the same time!!! We put the AKG D112 on the kick in channel 1. An SM57 on the snare into channel 2. We used the AKG 414's, cardioid setting, for spaced pair miking over the kit (RL), into channels 3 and 4. Spaced pair miking is cool because it aurally splits the kit into a right and left side, creating the opportunity for true stereo mixing with drums. The 3-to-1 rule is a good starting point to position the mics, but if it sounds like crap, it sounds like crap, and you should reposition until you hear what you like. Mid-side miking was the next technique we tried, using the AT 4050's into channels 5 and 8. The mid was set to cardioid, and the side - figure 8. Make sure to get those capsules as close together as possible! We used the Josephson e22's on the third set of overheads into channels 9 and 10, this time using X-Y miking. To explain this technique, from the drummers perspective, the left cardioid mic is pointing across and picking up the right side (approx 0 degrees axis at the ride cymbal). The right side OH picks up the left side (approx 0 degrees axis at the hi hats). I really like how this technique sounds, but the mic stands we used were too short too get the position we wanted. Next time we will use longer mic stands, or simply pick different mics for the job! After patching everything correctly we set up a session to get rockin'. Nadia played a groove while we listened, and recorded. To decode the mid-side, double the side track. Pan one hard R and the other hard L. Put one of the sides out of phase! Grouping tracks is a great idea for making mixing more organized and efficient. We grouped the kick and snare together, and separately grouped each of the miking techinques. This enabled us to A-B between the different overhead choices. The mid-side ended up sounding the fattest, it just picked up a great, full, warm low end sound. I was shown a little bit of mixing out of the box. Finally, no digital fader for a minute! We had to patch the mid-side tracks out of protools into the board, into LINE 1 inputs on the patch bay, channels 38 39 and 40. Now we were able to monitor from the faders right on the console. This week for our labs we decided it would be a good idea to start pre-production for our next song. We set up a couple of 414's as room mics, and put 2 Royer 121's on the piano. Myself on piano, Garret and Jake on guitar, Chris on drums, and Taylor in the studio as engineer. A few of us had brought in some ideas for a song, and a couple chord progressions. We jammed on them for a while and got a good half an hour of tracking in. We spend the rest of our lab time retracking our first efforts, because we had back up issues and performance issues.

Microphones!

This week we are learning the different types of microphones and their technical aspects. The 3 types of mics are dynamic, condenser, and ribbon. Dynamic mics work based on a magnetic moving coil (electro-magnetic). a copper coil moves around a magnet when sound pressure waves come in contact with the diaphragm (top of coil). Diaphragms can be large or small. This acts as a TRANSDUCER, which is converting the spw's into an electric current that goes down the mic cable into the board. Dynamic mics are less sensitive than condenser mics and can handle close micing better. Condenser mics need an additional power source, phantom power. This powers 48 volts to the mic, because these mics operate electro-statically. Ribbon mics use the electro-magnetic principle as well, and are very sensitive. Polar patterns are another important part of the anatomy of a mic. Some mics have varying patterns, in which case you can switch the polar lattern depending on the room. This is a way of picking up the sound, and there are different spherical shapes that respond differently. Cardioid is a very common pattern, and it is shaped like an upsidedown heart. A mic with this polar pattern will pick up noise best from the front and reject noise from the back. Figure-8 is another pattern, which will pick up sound from the front and back of the mic, and the null points are located on the sides of the mic. An omnidirectional pattern picks up sound from all sides, except the very back of the mic, because it is theoretically impossible to pick up with the body of the mic in the way. Super or Hypercardioid picks up like a cardioid, and has rejection at about 120 and 240 degrees. The rejection of sound from the mics is really just an attenuation of certain frequencies down a certain number of decibels. Frequency Response is how a mic picks up frequencies. Each mic has specific boosts and attenuations in frequency. Each microphone has an SPL or sound pressure level. Sound waves are the movement of air, so if sound hits a mic too hard it can cause the moving coil or ribbon to break, incapacitating the mic.

Back ups, and doing things the right way.

We totally screwed up, and things have been brought to the groups attention that we need to work on as we get to know how the studio works. Backing up our stuff seemed to be an issue only because it took so long to transfer information between devices. That problem is currently being worked on, we are getting a firewire adaptor! We will always be sure in the future to back up our sessions, because you never know when a system is going to get changed out or crash. We hit bad luck, some of the sessions we saved ended up not saving certain files, and we lost them. Things could have been okay had we asked about it then, and got some help. We didn't speak up, and tried to re-record the tracks we lost. Then we found ourselves to be running behind schedule on getting the project done, all because we hadn't backed up properly. We all realize how serious keeping files on back up is, and will make sure that we speak up if we have a problem in the future.

Josephson e22S

Originally invented for the recording needs of producer Steve Albini, the Josephson e22S is a small diaphragm condenser microphone with a cardioid polar pattern. The fact that this mic is a side address mic sets it apart from other condensers. Side address refers to the mic receiving sound from a perpendicular angle. The shape of this microphone allows it to be placed in a tight spot. This mic was also designed for spot miking, which is synonymous with close miking. It is a solidly built mic with a black satin chrome finish.
Specifications:
Directional Pattern: Cardioid
Frequency Response: 20-20,000 Hz
Rated Impedance: 200 ohms
Equivalent noise level: 15dB SPL A-weighted
Maximum SPL: 144 dB
Weight: 180 g
Dimensions: 21mm D, 190mm long, 30mm head

The e22s is an extremely versatile microphone and was made to handle high sound pressure levels. It was originally created for recording drums, but has been said to give a great response for many instruments. Steve Albini has used this microphone on drums (more specifically toms), electric bass and guitar. It has also been liked and used on acoustic guitars, mandolins, horns and organs. Examples of the e22S can be heard on just about any recording that Steve Albini has done, since he was quoted “Literally the only microphone used on every session”.

Josephson Engineering specializes in building condenser microphones for stage and studio settings, and field recording. The company was founded in San Jose, California in 1988 and later relocated to Santa Cruz. They also construct mic capsules for other leading companies as well. A unique thing about this company will build a custom microphone for you if cannot find the one you want.

Footnotes
http://www.soundonsound.com/sos/sep05/articles/albini.htm
http://www.josephson.com/
http://soniccircus.com/Josephson-e22S

dB measurement, Early Reflections and Reverb

There are several more terms that I became clear on this week. I used to think that decibels only measured the signal level of a sound source, but it also measures SPL, or sound pressure levels. If the distance is doubled from the source to the mic, there will be a 6dB cut in signal. I learned in class that we as humans are naturally programmed to hear a certain range of frequencies and we are more sensitive to certain frequencies. Humans are most sensitive to the frequencies of a crying baby. This is a built in survival tactic for humans.

Early reflections are a series of sounds, echoes, or frequencies that bounce off the surface of a room, and they are usually something that you don't want in the recording. These happen after the direct sound but before reverb actually starts. Reverb is different than early reflections in that it is the sound of a room. If early reflections combine, they can cancel out certain frequencies or increase in amplitude.

In the studio for our lab time, we have been tracking a song. This week we got bass and guitar tracked. We took the bass and went direct into the Millenia using a 4 string Ibanez. We did a few takes until we got down what we thought was good enough and fit with our time schedule. Everyone has been taking turns on the board and patching things in, so we are all becoming very familiar with how the process is working. For the guitar we decided to record out of the keyboard amplifier in room 100. We close miced the amp with an sm57, slightly off axis from the cone of the speaker. We decided to capture room sound as well so we set up a royer 122 about 6 feet from amp. I really liked how the royer sounded. It was much brighter than the sm57 for obvious reasons, the main one being a condenser vs a dynamic mic. Everyone has been having a fun but serious time in the studio, and we are all getting along!!

For our first project assignment we took two different microphones around to 8 different rooms, all with different acoustical characteristics. Everything went smoothly, I didn't seem to run into any difficulties. After getting our group of 5 people together, we set up a Protools session with 5 tracks. We had plenty of XLR cables, because we knew we'd be tracking in places quite far from the control room. The environments we chose to record in were MPA 100, the front foyer, the restroom, a hallway, outside, into a cement wall, into sonnex, facing a window, and off-axis. We used an sm57 and an AKG C414. The difference in sound from one mic to the other was quite drastic. The 414 was much more colorful than the sm57.

After tracking the different members in the group, we got to edit the tracks and listen back. Each person took turns editing their own tracks. Listening to my voice recorded in several different places became quite interesting and I was able to pick out characteristics of each. I thought MPA 100 was going to have a lot of reverb, but there turned out to be more echo than reverb. The sound was bouncing off several walls all over the place. A terrible room for recording if you are trying to record a close mic sound. The front foyer had a fair amount of reverb and echo as well, because of the tile floor and basically glass walls. The restroom had the most reverb, with a much longer tail, probably around 2-3 seconds. The hallway didn't sound as long as it is. When I was outside, as soon as I was done talking the sound was already gone because there were no walls to bounce off of. The sonnex was kind of the same, but I could tell just by listening to the acoustics that I was inside somewhere. Talking off axis, my voice sounded very faint, even though I was talking loudly. We used an omni-directional polar pattern on the 414, so off-axis wasn't needed.

We made sure to make several copies of our tracks, because audio doesn't exist until it is backed up at least two times! Everyone worked we'll together in the studio. I would say that the general problem is being on time. Throughout our busy lives, we do our best to be on time, but sometimes we are late.

Multi-tracking, Reverb send, Click track

Today we tracked a short one minute jam that involved recording drums, bass, an acoustic guitar, and vocals. We pretty much tracked the same way we did previously, with the AKG D112 on the kick, the SM57 on the snare, and the 4050's for the overheads. We went in channels 1-4 for the drums. When recording the bass, we when direct injection through the Millenia. Patching was different, however. We went in the patch panel on ch2 and into protools on ch5. Then we set up the SM57 for the guitar. Mic placement was off axis, and pointing at the 12th fret. Guitar was recorded on ch6 in Protools. We recorded vocals with the 4050. We then set up a reverb aux track for monitoring in the headphones. About a one second tail was used on the reverb. We bused the vocal track to 3-4 as well as 1-2. Set input on reverb track to bus 3-4. Setting up a click track was simply selecting TRACK - click track, in protools. Dont forget to change the sound of the click because the default is quite annoying. Marimba 2 seems to be the least offensive.

Tracking Drums and Monitoring Headphones

Today, drums were tracked in the studio. The four mics used were 1 SM57 on the snare, one Audix D112 for the kick, and 2 450's for the overheads. The patch panel inputs were Kick ch1, Snare ch2, OH R ch3, OHL ch4. A session was opened up in Protools and we started with 4 audio tracks. The sm57 was placed slightly off axis about 1.5 inches over the head from the rim of the snare drum. Spread Pair Miking was a technique we used for the overheads, which splits the kit in half, and a D112 was used to mic the kick drum inside. While a tech is playing the drums, watch signal levels and tweak the gain on the board mic pre to get the desired level of signal. Put 7 band EQ's on all 4 tracks, and solo the overheads and make sure they are not out of phase. To replicate a stereo sound for the kick drum, we then tried mid-side conversion miking. This requires 2 different cardioid patterns, cardioid (mid) and figure 8 (side). About 4 or 5 feet in front of the drums, place the figure 8 mic with one of the null points facing the kick. Place the cardioid pattern mic above and very close to the "side" mic, facing the kick. We need to get the stereo sound though! Double the side mic track. On the duplicate track, and an EQ and put it out of phase. Pan the original track hard left, and the duplicate track hard right. Now we need to get a headphones mix set up. Plug an XLR cable into chA on the patch panel and put the other into the out of the headphones box. Group the tracks together and create an aux track. Select all tracks, and in the sends area, send all tracks to BUS 1-2. On the AUX track input, select Bus 1-2 and on the output select interface B1-2. On the patch bay route B 1-2 into the samson. Also on the patch bay route headphones 1 out and plug into room 100A. You should now hear your mix. Listen for the difference in the mid-side conversion technique compared to miking inside the kick.

More audio fun...

Grab an XLR cable, mic, and let's hook it to a mic pre and get signal through ProTools! After plugging the cable into channel 1 on the patch panel, grab a patch cable and plug it into the mic line output 1. This bypasses the board. Put the other end into the NEVES input on the patch bay. Get another patch cable and plug it into the output of the NEVES and into Input A1 ProTools. Now go grab two more patch cables and put each one in ProTools output 1 and 2 on the patch bay. Plug the other two ends into 2Track (right and left). After opening a new ProTools session and saving it in a locatable folder, with the mouse record enable audio track 1. Check to make sure the audio input path selector reads "A1" and the output path selector reads (1-2). On the MTA 980, find the Aux Master track and set the monitor level to no higher than 9 o'clock. Make sure the 2KT1 and Mix buttons are depressed (on). You should now have signal through a mic pre and are ready to record!

That is what we did in class today. After getting the signal, various classmates went around to random parts of the music building and recorded audio, the beginnings of our first assignment. The purpose of this is to begin to hear how the shape of a room affects the audio recorded. Tomorrow my group goes into the studio, and then I will be able to hear the different acoustic qualities each environment has to offer.

1/25/10 - 1st Audio Production Class

Today was the first day of Audio Production class, and it was a good one. I met some new students, and saw some people that I know from last semester. In class we learned about what we are going to learn about, and we began signal flow today. We plugged an SM57 Shure microphone into the patch panel of MPA room 100. We got signal flow into ProTools and learned about the patch bay. I got to know what outboard gear we have available to work with and I saw where the cable and mic stand storage room was. The information presented in today's class was not all new to me, and I have already gained a clearer understanding of signal flow just by taking notes and listening/watching.