Category: My Audio Engineering Journal

This weeks theory focused on acoustics and it is going to be a bit more difficult to condense down into bite size chunks like I have in past blogs. It is also testing my understanding of the topic at hand to the maximum. So with that being said, let me go make a sandwhich cause I think we are in for a long read.
<a href="http://www.toaelectronics.com/images/img_spkr/Speaker_Software.jpg"><img style="FLOAT: left; MARGIN: 0px 10px 10px 0px; WIDTH: 320px; CURSOR: hand" alt="" src="http://www.toaelectronics.com/images/img_spkr/Speaker_Software.jpg" border="0" /></a>There was mention of a computer application that will calculate the ideal speaker locations for any given room. You simply input the room data, such as dimensions and furniture etc and let the program calculate the optimum speaker placement. However, due to the complexity and countless permutations it can take a week or two to generate the information. Yes, a week or two (you read that right). A specific application was not mentioned as we were talking about the application functionality in general. I did some hunting on the web and found this one. TOA Speaker System Design Software. I am downloading it for free from their website as I write this blog. I'll be checking it out and I'll post my review in a separate blog entry. <a href="http://www.toaelectronics.com/speaker_software.asp">http://www.toaelectronics.com/speaker_software.asp</a> UPDATE: I only had a chance to briefly check out this software package and I'm not sure that it will do what I described earlier. I will dig into this a little further at a different time.

We moved on to discussing "Room Modes" and why they are important to an Audio Engineer. A Room Mode Calculator can be found here <a href="http://www.mcsquared.com/metricmodes.htm">http://www.mcsquared.com/metricmodes.htm</a> Now then, I will try and explain room modes. A "Room Mode" is a set of frequencies that resonate between parallel surfaces. Room modes are calculated using the simple formula; frequency equals velocity divided by wavelength. For every 1000 hertz there are 9 modes. If you are like me you are asking yourself, why do I need to know about room modes? It is important to understand room modes as it will affect the quality of sound as the frequencies build up. Certain frequencies will get louder. The best example is to use the worst type of room, and that would be a cube. <img style="DISPLAY: block; MARGIN: 0px auto 10px; WIDTH: 358px; CURSOR: hand; TEXT-ALIGN: center" height="117" alt="" src="http://roommodes.com/image2.jpg" border="0" />In this example we will say that this cube room is 10' x 10' x 10' to make the math simple. Using the formula frequency=velocity divided by wavelength we can calculate the frequencies. We know that the wavelength is 10 feet and the velocity is 1100 so we can then tell that the frequency equals 110 hz. This figure increases by 110 hz so then you would have 110, 220, 330, 440, 550, 660 etc. However, in this example we can cut the 110 hz in half to 55 hz. 55 hz is equal to the A string on a bass guitar. So in that example room the A string would sound louder because the room will resonate the frequencies more and they pile up to make it louder. Now that the problem frequencies can be identified, we can move on to treating the room. More on treating rooms in a future blog.

<a href="http://www.cockos.com/reaper/images/reaper20-small.gif"><img style="FLOAT: left; MARGIN: 0px 10px 10px 0px; WIDTH: 320px; CURSOR: hand" alt="" src="http://www.cockos.com/reaper/images/reaper20-small.gif" border="0" /></a>Our session this week opened up with a discussion about file compression. This conversation was sparked by a recently created audio file compression discovered in Reaper called OGG. OGG files appear to be a very efficient means of file compression. Note: we were wondering about the OGG files being possibly the only option that Reaper would use, however I have looked into this and in fact Reaper supports many file formats. From the Reaper Help File

REAPER allows you to use different recording formats for different tracks in the same file. The default format for each new track that you create will be determined by your Project Settings. To change these or any track, simply right click over the track’s VU meter, then from the menu choose rack Recording Settings. You can choose any of the formats WAV, AIFF, Monkey’s Audio, FLAC,MP3, OGG Vorbis, or WavPack lossledd compressor. Depending on which format you choose, you will be presented with a range of options appropriate to that format.
<div>WAV files that have an original file size of 40 MB can be compressed to about 400 k throught the use of OGG files. This is approximately 1/100 of the original WAV file size. By comparison, a typical MP3 file will be 4 MB or 1/10 of the file size. Please note: the above ratios are yet to be confirmed but I will be searching the web for the exact ratios. The MTP compression that the Roland V-Studio uses compresses by approximately 40%.
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<div>Next we learnt about velocity (speed of sound) and effects. Sound travels at a speed of 1130 feet<a href="http://www.aviationexplorer.com/sonicboomplane_navy.jpg"><img style="FLOAT: right; MARGIN: 0px 0px 10px 10px; WIDTH: 293px; CURSOR: hand" height="205" alt="" src="http://www.aviationexplorer.com/sonicboomplane_navy.jpg" border="0" /></a> per second. For ease of calculation we are rounding down to 1100 feet per second. References of thunder clap we discussed as a way of determining distance. Temperature also enters this equation. For every drop of 1 degree, sound will slow down by 1 foot per second. Here are some notes about effects. Reverb can be referred to as the overall ambient characteristic of the sound. Decay time refers to how long the note/sound sustains. Typical examples are a 2.5 second decay at a rock concert, .4 second decay in our classes live room, and 6 seconds in a cathedral. Diffusion refers to how smooth the reverb is. Early reflections are discernable echo/series of repeats. Repeats must be more than 35 milleseconds before a destict repeat can be detected. Chorus modulation is at a fixed rate and decay.</div>
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<div><a href="http://img.musiciansfriend.com/dbase/pics/products/tn/8/9/2/245892.jpg"><img style="FLOAT: left; MARGIN: 0px 10px 10px 0px; WIDTH: 36px; CURSOR: hand; HEIGHT: 89px" height="689" alt="" src="http://img.musiciansfriend.com/dbase/pics/products/tn/8/9/2/245892.jpg" border="0" /></a>Now for the fun stuff. We began recording one of the instructors original tunes. In this session we first recorded a scratch track. This was done by recording a rough version of the song using one microphone and was set to a click track at 99 bpm. In this case we used an AKG 414 set to a circle pattern. The placement of the mic allowed for both the vocals and guitar to be well balanced. The location was approximately half way between the sound hole of the guitar and the mouth of the performer and about 12" in front. We then began replacing the scratch guitar parts with overdubs. One guitar overdub was panned hard left while a additional guitar overdub was panned hard right. Once we were happy with the guitar parts, we moved on to recording the vocals. We only had enough time to lay down one vocal track but the quality and performance were good enough to print and move forward. We saved the project and will be adding more instruments in the next lesson. </div>

<a href="http://bp2.blogger.com/_KcCGgmBQFy4/R7kAN0bctXI/AAAAAAAAAFg/ihZcRGY_h4o/s1600-h/IMG_0600.JPG"><img id="BLOGGER_PHOTO_ID_5168162284796425586" style="FLOAT: left; MARGIN: 0px 10px 10px 0px; WIDTH: 148px; CURSOR: hand; HEIGHT: 176px" height="250" alt="" src="http://bp2.blogger.com/_KcCGgmBQFy4/R7kAN0bctXI/AAAAAAAAAFg/ihZcRGY_h4o/s320/IMG_0600.JPG" width="207" border="0" /></a> I have been a recording and creating music for the past 6 years. I am mostly self taught with the help of many great sites on the internet. I am in a band called The London Project and I record all of our stuff in my home studio. You can listen to our latest music at <a href="http://www.thelondonproject.ca/">http://www.thelondonproject.ca/</a>

I have been interested in music all my life. I guess what started it all was a Christmas present when I was 9 (35 years ago). A cassette recorder and 3 tapes; Nazereth Rampant, Gordon Lightfoots Greatest Hits and The Stampeders. It came with a couple of blank tapes as well as a microphone. Didn't take long before I was recording my voice doing hockey broadcast impersonations.

A few years later I began collecting LP's. I still have most of them (about 500 in total). I eventually used my knowledge of music to get me some DJ work. Spun the platters just for fun at company parties etc. while going to college to learn Graphic Design. My graphic knowledge launched a career in the Trade Show and Signage industry. Working daily on computers lead me down the Information Technology path which is what I currently do for a living.

I selected this particular course as it was in the evenings and would allow me to keep working.

The course is broken down into six levels. The first three levels are grouped together and teach you recording fundamentals. Rather than listing what the literature says about what we will be doing, I will write separate blogs on what we were actually taught.

I hope you find these blogs helpfull. I'm not the greatest writer but I will try to keep you entertained.

I left home early so that I would have plenty of time to check things out at the college (never been there before). Parking wasn't a problem at all. (due to the fact that this is an evening course I'm sure). So off I go into the school. Wow, its big. I see that there is a large lobby with 3 wings. Cool cause I know that it is in the 3rd wing. Signs are everywhere to guide me. Room 3825 is just over there through the doors on the right, I say to myself. Found the room pretty quickly but it started to feel kinda strange. There is no studio here and why are there a bunch of people talking <a href="http://bp3.blogger.com/_KcCGgmBQFy4/R5DH6DBBZ4I/AAAAAAAAAA0/VtPCSSLUFr4/s1600-h/DSC01117.JPG"><img id="BLOGGER_PHOTO_ID_5156841373395150722" style="FLOAT: right; MARGIN: 0px 0px 10px 10px; CURSOR: hand" alt="" src="http://bp3.blogger.com/_KcCGgmBQFy4/R5DH6DBBZ4I/AAAAAAAAAA0/VtPCSSLUFr4/s320/DSC01117.JPG" border="0" /></a>about physics. Hmmm... Back to the front lobby I go. Quickly spotted the general information area and found out the guy gave me the WRONG ROOM NUMBER! It was actually 3285. Now it is getting close to the time the class starts and I still don't know where the room is. Well, eventually I found the room with about 5 minutes to spare. Two students were already there. Four more arrived to make a total of 7 in our class.
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<div>We grab some chairs and start our session in the live room.

<a href="http://bp0.blogger.com/_KcCGgmBQFy4/R5DG_TBBZ3I/AAAAAAAAAAs/Z1NKJ7t25Mw/s1600-h/DSC01119.JPG"><img id="BLOGGER_PHOTO_ID_5156840364077836146" style="FLOAT: left; MARGIN: 0px 10px 10px 0px; CURSOR: hand" alt="" src="http://bp0.blogger.com/_KcCGgmBQFy4/R5DG_TBBZ3I/AAAAAAAAAAs/Z1NKJ7t25Mw/s320/DSC01119.JPG" border="0" /></a>The teacher introduced himself and asked a couple of basic questions like, how did you hear about us etc. He moved onto the session pretty much right away. He showed us some cable types and began talking about what it takes to make a good recording. "Good musician+good room+good instrument+good engineer=proffessional product". "But how do you define a good instrument", he says. He then used drums as an example. How do you know a good drum kit? By the way it sounds, and general reputation. He went on to say that a good drummer can make a crappy kit sound good.

On to learning about SPL and decibels. Here is the basic breakdown;
-To increase the volume units (VU) by 3 dB, 10x the energy is required. To increase by another 3 dB, 20x required. To increase yet another 3 dB to make a total of a 9 dB increase requires 40x the energy etc. It is an exponential scale.

Here are some quick info bits on dB.
74 dB is conversation level. This is based on the phone system. It was a comfortable volume to speak to someone on the other end of a phone.
118 dB is the threshold of feeling.
130-140 dB is the threshold of pain
Ambient noise is around 25-35 dB.

<a href="http://bp3.blogger.com/_KcCGgmBQFy4/R5DI5DBBZ5I/AAAAAAAAAA8/gze4Jo8ZGck/s1600-h/DSC01116.JPG"><img id="BLOGGER_PHOTO_ID_5156842455726909330" style="FLOAT: left; MARGIN: 0px 10px 10px 0px; CURSOR: hand" alt="" src="http://bp3.blogger.com/_KcCGgmBQFy4/R5DI5DBBZ5I/AAAAAAAAAA8/gze4Jo8ZGck/s320/DSC01116.JPG" border="0" /></a>After a quick break we went into the control room. He introduced us to the equipment which included a Peavey 1600 32 channel console, Avalon pres, Focusrite, Tascam and some really nice vintage Tannoy monitors and a set of Genelecs. A Roland DAW was hooked up to a set or Roland monitors as well. Too much stuff to list so I will try and get some pictures next week.</div>
<div>EDIT: These pictures were taken during the second session but I thought it would be appropriate to update this post rather than start a new one.</div></div></div>

<a href="http://bp3.blogger.com/_KcCGgmBQFy4/R5TCbDBBZ8I/AAAAAAAAABY/Aq7ZmRp6JQE/s1600-h/DSC01111.JPG"><img id="BLOGGER_PHOTO_ID_5157961243167909826" style="FLOAT: left; MARGIN: 0px 10px 10px 0px; CURSOR: hand" alt="" src="http://bp3.blogger.com/_KcCGgmBQFy4/R5TCbDBBZ8I/AAAAAAAAABY/Aq7ZmRp6JQE/s320/DSC01111.JPG" border="0" /></a>I got lost again! Well, not really but still don't know my way around the school. I ended up being one of the first to arrive. I got there in time to see some band members loading up our isolation booth with instruments!! Band dudes! Get out of our room. I didn't say it but I sure was thinking it.

<a href="http://bp1.blogger.com/_KcCGgmBQFy4/R5DKrjBBZ6I/AAAAAAAAABE/fevpi3V_7O8/s1600-h/DSC01112.JPG"><img id="BLOGGER_PHOTO_ID_5156844422821930914" style="FLOAT: right; MARGIN: 0px 0px 10px 10px; CURSOR: hand" alt="" src="http://bp1.blogger.com/_KcCGgmBQFy4/R5DKrjBBZ6I/AAAAAAAAABE/fevpi3V_7O8/s320/DSC01112.JPG" border="0" /></a>The instructor arrived right on time and after a bit of chit chat he talked about one of his students that brought in his SAW system. Built with a Asus MB, SATA drives, Lightpipe and a Frontier interface. We didn't get a chance to see it or discuss it much more. Sounded interesting and I'd like to learn more about it.

We checked out a ProTools system. It was an older 001 running with PT 5.01. Quickly went over the interface. 2 mic pres, and 6 analogue ins. We opened up a project and did a basic mixing
overview.

<a href="http://bp1.blogger.com/_KcCGgmBQFy4/R5BiejBBZ0I/AAAAAAAAAAU/i7sWiGDfw3M/s1600-h/DSC01114.JPG"><img id="BLOGGER_PHOTO_ID_5156729850274342722" style="FLOAT: left; MARGIN: 0px 10px 10px 0px; CURSOR: hand" alt="" src="http://bp1.blogger.com/_KcCGgmBQFy4/R5BiejBBZ0I/AAAAAAAAAAU/i7sWiGDfw3M/s320/DSC01114.JPG" border="0" /></a>Into the control room we go. We spend a bit more time on the board this time. First off we learnt a bit about the board itself. This is a 32x16x32 board meaning, 32 input channels, 16 sub channels and 32 monitor channels.

We then went on to reading the meters. The meters read voltage with 1.23 volts = 0 on the meters. This is considered pro standard. 0 being optimum. In comparison, VU meters on a consumer cassette deck only require 0.316 volts to be at optimum. Because of this variation, it is important to watch that you don't overdrive consumer level products with the board. He went on to say that broadcast standard requires 1.96 volts = 0.

We also found out that 0 on the board is equal to -20 on the Tascams. Another way to look at it is, 0 = full scale on a digital syatem which equals +20 on the board.

<a href="http://bp2.blogger.com/_KcCGgmBQFy4/R5BqXzBBZ1I/AAAAAAAAAAc/NaHjZTb9mog/s1600-h/DSC01113.JPG"><img id="BLOGGER_PHOTO_ID_5156738530403247954" style="FLOAT: right; MARGIN: 0px 0px 10px 10px; CURSOR: hand" alt="" src="http://bp2.blogger.com/_KcCGgmBQFy4/R5BqXzBBZ1I/AAAAAAAAAAc/NaHjZTb9mog/s320/DSC01113.JPG" border="0" /></a>Next up... the Roland V-Studio 2480. This is a DAW unit with 8 preamps and 16 inputs. It will record up to 96k - 24 bit. It can be connected to a pc through the use of Roland R-bus PCI cards (up to 2). It has a built in CD burner and a 80 gig drive partitioned into 6 drives. It uses native compression (MTP) to save space on the drive.

AT this point there would be way too much to describe in this blog as we dove into the DAW quite a bit. If you want to learn more about the Roland you are going to have to RTFM. Basically it was pointed out that we are going to be using this piece of equipment at lot in this semester.

I guess that is about it for this week. Bye for now.

Heyyy!! I didn't get lost. Not only didn't I get lost but I confidently walked straight to the class. This was really good news for me as I was running late and was quite worried that I would be late for class. We started this session out by burning up some song tracks to CD so that everyone had some material to work with. They were burnt at 4x so that it would be a nice deep burn.

While the CD's were burning we entered a discussion about analog versus digital. Upon discussion we quickly realised that even though we all had a basic idea of what the two technologies were, we were not able to quickly describe them. In analog recording systems, the continuously varying amplitude of the sound waveform is <a href="http://bp0.blogger.com/_KcCGgmBQFy4/R5oYP0mBG1I/AAAAAAAAACc/maR3MIrZskc/s1600-h/principles_book_cover.jpg"></a>translated to a continuously varying level of magnetism, LP groove amplitude, etc. In contrast, with digital audio encoding, <a href="http://bp0.blogger.com/_KcCGgmBQFy4/R5ona0mBG4I/AAAAAAAAAC0/iMgARIE2VNg/s1600-h/principles_book_cover.jpg"><img id="BLOGGER_PHOTO_ID_5159479664854506370" style="FLOAT: left; MARGIN: 0px 10px 10px 0px; WIDTH: 164px; CURSOR: hand; HEIGHT: 188px" height="153" alt="" src="http://bp0.blogger.com/_KcCGgmBQFy4/R5ona0mBG4I/AAAAAAAAAC0/iMgARIE2VNg/s320/principles_book_cover.jpg" width="145" border="0" /></a>discrete (noncontinuous) time sampling and amplitude quantization is required, a process that breaks the originally continuous and smooth waveform into a staircase or other pattern of pulses. The instructor went on to mention that "The Principles of Digital Audio by Ken Pohlmann" was a really good read for anyone that wants to get in depth information about digital audio and it's origins. I found a summary of this book here <a href="http://mcgoodwin.net/digitalaudio/digitalaudio.html">http://mcgoodwin.net/digitalaudio/digitalaudio.html</a>.

After a short break we began discussing a bit about frequencies of instruments and why it is important to know them. Basically it is important to know the frequencies so that through the use of EQ you can increase (amplify) or decrease (attenuate) the corresponding frequency allowing for the instruments to sound better in the mix. We briefly talked about impedance which is basically resistance. High impedance cables are intended for short distance of under 15 feet. By contrast low <a href="http://bp3.blogger.com/_KcCGgmBQFy4/R5onAkmBG3I/AAAAAAAAACs/e3qUA5gn1BE/s1600-h/mic_closet.JPG"><img id="BLOGGER_PHOTO_ID_5159479213882940274" style="FLOAT: right; MARGIN: 0px 0px 10px 10px; CURSOR: hand" alt="" src="http://bp3.blogger.com/_KcCGgmBQFy4/R5onAkmBG3I/AAAAAAAAACs/e3qUA5gn1BE/s320/mic_closet.JPG" border="0" /></a>impedance cables allow for an audio signal to travel a lot further without <a href="http://bp3.blogger.com/_KcCGgmBQFy4/R5oiQkmBG2I/AAAAAAAAACk/J7_2Vl_3FkA/s1600-h/mic_closet.JPG"></a>introducing signal loss. It is important to select the appropriate cable for the task at hand.

"Enough of the theory stuff" says the instructor and off to the control room we go. We begin with a quick review of the Roland DAW. When quickly moved on to the steps required to route a signal to the Roland DAW and onwards to the Peavey console. We setup the a computer to loop a song for continuous playback which would be used as a quick and easy sound source. We grabbed a SM57 from the mic closed and set it up about 12" directly in front of the computer. We then plugged it into the closest wall plate. Once this was done we returned to the control room to route the signal into the DAW and console. Once all the routing was physically connected we began working with the Roland DAW routing so that we could forward the signal to the console. I will stop writing here as there would be too many steps to list.

In this blog I would like to welcome my fellow AE students! Hopefully these blogs will help you as much as they help me. I should point out however that this blog is primarily written to re-enforce my understanding of the AE sessions. Certain information may be missing as I may have already gained this knowledge. Therefore it is important to say that this blog should not be a substitute for your own notes but rather a supplement to them. <div><div><div><div><div>
<div></div><img id="BLOGGER_PHOTO_ID_5162063109157886930" style="FLOAT: right; MARGIN: 0px 0px 10px 10px; WIDTH: 178px; CURSOR: hand; HEIGHT: 95px" height="127" alt="" src="http://bp2.blogger.com/_KcCGgmBQFy4/R6NVDEmBG9I/AAAAAAAAADc/wVff13e09f0/s320/waveform.gif" width="207" border="0" />This weeks session began with a discussion on WAV files. We took a look at a variable waveform in ProTools. Zooming in on the wav <a href="http://bp3.blogger.com/_KcCGgmBQFy4/R6NU1UmBG8I/AAAAAAAAADU/PHkoLLRy_eI/s1600-h/r2rdac_waveform.png"><img id="BLOGGER_PHOTO_ID_5162062872934685634" style="FLOAT: left; MARGIN: 0px 10px 10px 0px; WIDTH: 137px; CURSOR: hand; HEIGHT: 87px" height="130" alt="" src="http://bp3.blogger.com/_KcCGgmBQFy4/R6NU1UmBG8I/AAAAAAAAADU/PHkoLLRy_eI/s320/r2rdac_waveform.png" width="202" border="0" /></a>file we were able to see the peaks and valleys. It was pointed out to us that the waveform includes both amplitude and frequency information. The image to the right displays a typical waveform as viewed in many audio software applications. The image to the left displays a zoomed in image of the waveform. The jagged edge displays the frequency while the overall shape of the wave represents the amplitude.
<div></div><a href="http://bp3.blogger.com/_KcCGgmBQFy4/R6NYFUmBG-I/AAAAAAAAADk/vyDN73nXOwI/s1600-h/mic-diaphragm.gif"><img id="BLOGGER_PHOTO_ID_5162066446347475938" style="FLOAT: right; MARGIN: 0px 0px 10px 10px; CURSOR: hand" height="107" alt="" src="http://bp3.blogger.com/_KcCGgmBQFy4/R6NYFUmBG-I/AAAAAAAAADk/vyDN73nXOwI/s320/mic-diaphragm.gif" width="261" border="0" /></a>
<div>Next up in our session was to learn about microphones and how a microphone operates. The example we looked at was a Shure SM58 dynamic microphone which is a staple in most every studio. The SM58 is known for it's ruggedness and it's ability to perform well on stage.</div><a href="http://bp2.blogger.com/_KcCGgmBQFy4/R6NZzEmBG_I/AAAAAAAAADs/90NQbbnOtZo/s1600-h/mic-dynamic.gif"><img id="BLOGGER_PHOTO_ID_5162068331838118898" style="FLOAT: left; MARGIN: 0px 10px 10px 0px; CURSOR: hand" height="200" alt="" src="http://bp2.blogger.com/_KcCGgmBQFy4/R6NZzEmBG_I/AAAAAAAAADs/90NQbbnOtZo/s320/mic-dynamic.gif" width="265" border="0" /></a>
<div>Microphones are a type of transducer which is a device that converts energy from one form or another. Microphones convert acoustical energy (sound waves) into electrical energy (the audio signal).
Different types of microphone have different ways of converting energy but they all share one thing in common: The diaphragm. This is a thin piece of material (such as paper, plastic or aluminium) which vibrates when it is struck by sound waves. In a typical hand-held mic like the one below, the diaphragm is located in the head of the microphone. When a magnet is moved near a coil of wire, an electrical current is generated in the wire. Using this electromagnetic principle, the dynamic microphone uses a wire coil and magnet to create an audio signal. The diaphragm is attached to the coil. When the diaphragm vibrates in response to incoming sound waves, the coil moves backwards and forwards past the magnet. This creates a current in the coil which is channeled from the microphone along wires. </div><div></div><div></div><div>Loudspeakers perform the opposite function of microphones by converting electrical energy into sound waves. This is demonstrated perfectly in the dynamic microphone which is basically a loudspeaker in reverse. When you see a cross-section of a speaker you'll see the similarity with the diagram above. In fact, some intercom systems use the speaker as a microphone. You can also demonstrate this effect by plugging a microphone into the headphone output of your stereo, although we don't recommend it! I have an old speaker that I have wired up to be used as a kick drum microphone and it works quite well in that application.</div><div></div><div>Well folks, that is about it for now. I will update this post at a later time. Discussion on the DI Box, gain staging and a look at the Shure KSM32 will be added at a later time.</div><div></div><div></div><div>*REMINDER: There will not be a session next week.</div></div></div></div></div></div>