Sound Design

Sound Design - Task

4 original voice patches and sequence them.

1 - Produce a fat bass, the voice patchess for subtractor zyp file format.

Rules only main wave forms.

ADSR - Duff duff duff - short attack, or dooooossssh longer release.

2 - Produce a Bright bell sound

Shape of the sound refelected in the wave you use. A bell would be a sharp sound, triangle can suit but need boosting. Piiiiiiing -  more release or more sustain. Ping - short release

3 - Produce a chromatic percussion- drum sound with a pitch, xylophone, wood blocks, percusive and chromatic.

4 - Produce a Piano/organ.

Create four subtractor synths, with a 16 bar loop with all the sounds. 


Subtactor-

Anologue synthesiser but its not really its digital.

Live Sound

Live Sound 

Live sound is a really challenging lesson. Sound and production skills is very technical and hands on with the hardware where as live sound is very hands on and focused more on application of our knowledge. 

Todays lesson we had the challenge of setting up a mixer, a mic, a foldback monitor, line level instrument with di box, and fx unit. 

We had to use prior knowledge in order to accomplish this task. 

 DI stands for direct injection it's used for line instruments that have a lower output then a mic instrument. 

Line is quieter and samplers, guitars, bass, keyboards and kaos pads are examples of line instruments. 

In the 60s/70s DI boxs where uused to boost line level signal to match that of the microphone so we can mix with mic and get a balanced mix. DI boosts the output. 

I has an earth switch used to cut out hums in the studio. 

Pad switch/+4+6 - if signal isn't boosted enough or is boosted too much then a pad can be used to reduce the signal or a +4/+6 boost can be used to increase. 

Most are mono but some are capable of stereo out. 

DI boxs have jack inputs, and xlr inputs and outputs.

sound production and studio skills

Sound Production and Studio Skills 

28th September 2012

Today we went round each of the three studios with the task of setting up a microphone and headphone amp in each, record a sound and play it back through the monitors and head phone amp.We had a limit of 45 minutes per studio. The analogue studio I'm getting quite familiar with. I felt most confident setting this studio up and troubleshooting any problems. In the digital studio it was quite easy to set up I'm slightly less confident in this studio as there seems to be a lot to set up and learn on the desk. The Command unit seems to be quite easy to set up the mixing surface is actually a control surface specifically designed for use with logic. The pre amp is on a rack next to the surface and is used to set gain and select channels (i think the input channel). I need more time in this studio as I have only been in it once.

As each studio has different hardware each one has it's own challenges. 

Command Unit                                

Has a control surface

not a mixer

Bus 7/8 - 

To send audio back to 

wall plate needs to be output

in logic on bus 7/8

If you restart the mac

you cannot log on

hold power button 

and keep pressed

the preamp is a rack 

on the side of

the desk

Mackie

The mackie desk has

two banks

1-24 which is to record

and 25 - 48 which

is for playback

9/10 aux phones

fat channel (additional side monitor) = 

check sound is 

routed to logic

select 'main' 'master l+r'

Analogue

Theres an on switch for each channel

Groups + Buses

each channel has a group to send

press 1-2 to send to bus 1-2 pan l+r

because it sends out and in to tape 

machine

'mix' button is for monitoring sound

mix output (main volume fader)

copied and seent to headphones

Class

We worked through each room setting up a mic, stand, headphone amp, logic, recording a track and playing back in each studio. I hadn't been in the command studio before so there where a few teething problems as I was unaware of the pre amp rack, hadn't used this type of control surface before, and hadn't had hands on experience setting up the buses. With Robs help got to understand it very quickly. The analogue studio I've had a good bit of experience with, and the digital mackie I've only had a brief amount of experience with.

I need to get some more experience in the digital studio as I'm a little unsure of the bank selects and sending sound out of logic through the desk.

We talked about upcoming tasks we will have to learn in mic technique

- Vocal and percussion

- electric and bass guitar with amp

- electric and bass guitar with DI box

- Keyborad - recording the audio while also recording the midi

Basic mix:

A band wiith a drumkit, mixing is core in unit 25.

Reason - Getting Started with sound creation

We've started looking at how to use Reason. I think this is just because of it's ease of use and the ability to be able to look at the back end of the simulated hardware and change the routing by moving around the cables.

When you load Reason your give it's default screen. For our purposes we want to set up a blank starting point.

In the preferences we need to change the default song to empty rack.

We then open a new track.

First we create a 14:2 mixer, thats 14 mono channels and 2 stereo channels.

Next we create a Subtractor synth

We also need to set up out keyboard so we can play notes back. 

In order to do this we go into preferences again 

Where it says general we select keyboards.

]

To connect the keyboard firstly click add.

You can select Manufacture under the drop down it will give a few different brands, you can also click on <other> and find it automatically. 

Once you've found your keyboard controller and added it correctly press ok and exit back into the main Reason screen. You should be able to press a key down and it will play back a sound. If it's the default pre set on subtractor you should get a bass sound playing.

Subtractor has pre sets built in for different sounds, these are sounds some one else has created and designed. If we change the settings of these patches we are manipulating the sound, if we use the pre set we're using someone elses work. 

We want to learn to make sounds from scratch which will be entirely unique to for us to use. So we initialize the patch, ie reset everything. 

No we will have a completely fresh starting point. 

Sound Creation

Wave Forms:

We discused the four main wave forms Sine, Saw, Triangle and Square. 

Sine has little or no harmonic content.

Square has a lot of harmonic content often used for synthesis, bass.

Triangle has little harmonic content

Saw/Ramp has a lot of harmonic content and is also often used.

ADSR

We also talked about ADSR. It stands for Attack, Decay, Sustain and Release. The envelope changes the way sounds triggered.

When we activate a note, the attack determines how long it takes for the sound to start from when the note is pressed. A short attack would mean the note would sound instantly a long attack would mean it would take longer for it to reach the highest level set. Decay is the amount the sound drops after the initial hit. A high level decay and long decay the note would sustain at the highest volume. If you dropped the decays level below that of the maximum attack the sound would at first sound loud and drop down quieter while the note is pressed. The sustain is how long the note is held and the release is how quickly the sound attenuates to unity gain A long release would mean the sound would continue after being released where as a short release would mean it stop instantly. 

Other Factors

Pitch, tone and volume also play a part in sound design but we haven't started to look at that. Being able to adjust a sounds ADSR you can create many different sounds and effects using any simple waveform. 

We adjusted the ADSR to see there effects in the amp envelope of subtractor to hear the dfferences in sound.

Future Assignments:

We will have to make 4 seperate sound patchs for our first assignment. 

A simple bell sound, no effects, quick attack, a singular bell ring.

A sound that is of chromatic percuasion a sound that has no pitch to it/ Xylophone or wood blocks would be an example. 

A slow lazy bass.

And an Organ - which is more difficult ro make.

This willl in a 16 bar loop format. 4 track of each sound.

Sound Production and Music Recording Skills Part 2

Sound Production and Music Recording

Part 2 - Microphones

We spent the last part of the lesson looking at microphones, their specifications and their uses. There are many different specifications that effect how a microphone performs. If we know the specifications for how different mics work we can choose the best microphone for a job and later as we learn the sound of different mics be able to choose which ones we would like to use. 

Design Type:

Dynamics - More robust, durable, can take a beating, can handle loud sounds.

The SM58 is the "industry" standard dynamic mic good for live or studio

Can be bought for around £60

Capacitor/Condensor - easily broken, more sensitive more detailed sound. Require Phantom Power of 48v.

A Se se2200a condensor mic (a good first mic)

Can be bought for about £150

.Valve - A lot more expensive, very sensitive, create a smoother sound.

Se Gemini 2

Can be bought for about £845

Sensitivity - This is how easily the mic can pick up quiet sounds. Some aren't very sensitive as they are designed to capture very loud sounds such as a kick drum. 

AKG D112 Dyanmic mic. The "egg" designed for kick drums.

Frequency Response - This is how well a mic will pick up different frequencys , usually from 20Hz to 20,000Hz (range of human hearing). Some mics can operate beyond this range. Some cannot cover the whole range of human hearing. 

Polar Patterns - Defines how well a mic picks up sound from different directions. Some Mics have switchable polar patterns. 

Omni mics can capture multiple singers in one mic, or be used to capture reverb, Cardoid capture from the front of the mic only which reduces spill in a loud environment, Bi directional captures both sides and can be used for two vocalists and mid side techniuqe.

Buttons/Switchs - These can vary performance and include:

a) On/Off (cheap mics)

b) Polar Patter

c) PAD button - Makes the mic less sensitive to low frequencys and can create a boomy effect in vocal performance. (40/80 cut)

e) Presence Switch - Boosts response in higher frequency range (2-8KHZ) to add presence to what is being miked. 

Quality - Mics range in quality and the components can effect the sound captured. Top end mics like Neumann can cost £1500 or more. 

Sound Production and Music Recording Skills Part 1

Sound Production and Music Recording Skills

Lesson 1 - Monday 10/9/2012

Part 1 - Headphone Amp and Monitoring Playback.

We started todays lesson by talking about headphone amps and reasons for their uses. A headphone amp is used to make the sound coming from the control room to the live room louder, boosting the signal so that it can be split between multiple headphones.

Every studio will have small or large differences in equipment, design and space. Being aware of this will enable one to be flexible and quickly adapt to new situations without having moments of panic and not knowing whats going wrong.

Following up from the diagram last week of signal path in order for the headphone amp to be connected the sound needs to be routed from the control room back into the live room via the wall plate.

We were show 2 set up diagrams for (i think) a digital and analogue control room, to show the signal flow from the control room to the wallplate (picture above)

102 Command -

106 Analogue

(spelling error should read analogue)

Practical Exercise

Today our practical exercise was to set up a microphone for recording and monitoring, with knowledge of the signal flow through the patch bay, mixer, into the mac/pc running logic, and the signal flow back into the room. We also started using the patching bay to change the signal flow into different channels. 

For referencing purposes I have made a diagram cutting down the equipment to the bare essentials for this task in order to make following whats going on a lot easier.

The "essentials" for this task.

A vocalist would sing into the microphone, the microphone converts the acoustic sound energy into electrical sound energy, which travels down the xlr balanced cable to the snake box. A balanced lead will enable hum and buzzes to be eliminated, we touched on the fact one of the L or R is inverted phase (which we will cover later) is the reason this is possible. The 3 pin xlr cable then goes into the snake boxes first input, which is connected to the wall box which, in turn, is connected to the patch bay. 

Snake box - Input 1 is broken/doesn't work and we had problems

with input 2, input 3 was what we found to a reliable input for us to rely on.

In a real studio situation some initial problems usually occur when trying to get a sound in the control room or mac/pc. For example the microphone might not work, cables by being twisted and wrapped up can be damaged as inside it is copper which can be broken. Some inputs might be broke, pins on xlrs can break a myriad of problems can occur in any sound situation and sometimes it can be even get to the point where recording sessions can get cancelled.

Now the patch bay in this studio is set up so that the top row of inputs is connected to bottom row of the inputs at the back, this is true for every plug in the patchbay except for the blue labelled ones.. The top row of inputs(which is yellow in this studio) are numbered 1-24 to match all the inputs on the snake box. The bottom output row (also yellow) are numbered 1 -24 and go to the input channels on the mixing desk into inputs 1 - 24.  

Patch bay with a cable redirecting the signal from input 3 to output

4 on the mixing desk. You can also see the blue sections which aren't connected top to bottom.

The mighty mixing desk.

Problems can occur at any point during the signals flow. Considering the patch bay has many many cables if you could have a manager in a studio who at the end of the night might unplug or change the patching in the bay which would lead to major problems with getting the signal flow to the mixing desk!

In this analogue studio the sound is not only patched into the input of the desk, but each channel then outputs to a reel to reel machine and comes back into the same channel to enable the use of this lovely piece of equipment to capture that tape sound and still use our fancy hardware and digital tools. 

Close up of channels on mixing desk.

This row of 5 buttons on each channel is very important. 

The MIX button at the top enables or disables the ability to monitor the sound through the amp and speakers in the control room. If theres no sound coming out the amp and speakers, in this studio, it's possible the mix button isn't pressed.

The next four numbers are the bus buttons. Pressing one of these will send the signal to the bus on the mixer, now for this mixer it's essential as I mentioned above there is a reel to reel machine in this mix. In order to send the signal to the mac/pc you must press a button down to send to the bus and raise the volume of the bus to get a signal to the software. If the sound isn't coming from the channel into the bus then it's possible you might have the wrong button pressed, it may be panned to the wrong channel or the gain might not be set high enough.

There is a pan control above these buttons, if it's set in the middle the signal will go to both Left and Right and the 1-2 button is pressed the signal will go to channels 1 and 2 on the bus. If it's panned right it will go to bus 2 and if panned left it will go bus 1. From the bus 1 output it will go to the input 1 of the AD/DA interface. Bus 2 output would go to bus 2 of the AD/DA interace and so on. If the 3-4 button was pressed the same principle applies, depending on panning will determine either bus 3 or 4 and the output would go to the corresponding AD/DA interface input. If theres no sound travelling from the bus to the computer it might be that the wrong input is selected on the daw which needs to match the AD/DA output, which if the sound is coming into bus 3 it would be going out into channel 3 on the AD/DA input.

Bus/Sends - This is where the signal comes from the channel and 

into the AD/DA interface due to the analogue nature of this 

study.

On the computer side of things we have logic loaded up with an empty template. We press the plus button on the top left where the tracks are usually and this creates a new track. In order to record we need the enable record checked and we need to select the channel input to match up with the bus the sound is coming from.

You would set the channel fader volume to 0, the bus level to 0 and adjust the gain on the channel until at the performers loudest performance there was no peaking. Maybe at 12 o'clock maybe a little more. 

DA half rack unit turned on

The mix out from the mixer travels to the half rack sized DA input/output and goes to the wall plate in the live room. If this isn't turned on no signal will be sent to the wall plate and neither the headphone amp or the headphones will pick it up. 

wall plate with xlr to jack cable

The wallplate connects to the headphone amp with an xlr to jack cable. The headphone amp is essential for live monitoring as many people might need to be listening at the same time to the mix being played. If it was a band for example monitoring can be highly useful. A guitarist might want to be able to hear a clean and distorted tone, a vocalist might need to overdub the vocals after the musics recorded having closed headphones (to prevent sound leakage into the mic) would enable them to sing in time with the song. Instruments might need re recording and this would also enable them to re record in time with the song. 

The musicians might want to hear different mixes while playing together, a guitarist might want his guitar louder so he can hear what he's doing, a vocalist might not want to hear his vocals while singing (or he might want to). These are possible without effecting the final mix (we haven't touched on this in depth yet).

Headphone Amp on a Piano

The headphone amp was connected with a IEC power to the wall power socket. The headphones are plugged into the output. 

Now if all is set up correctly we should be able to sing in to the microphone, hear a sound throught the speakers in the control room, record in to logic, listen to whats being recorded/played through the headphones in the live room. 

Hypothetically let's talk this through. We have the mic plugged into the snake input 1, this is coming through the patch bay to the mixer. We have a drummer playing and can hear him over the monitors. We press the 1-2 button, pan the channel left and we have the signal coming through the bus into the AD/DA interface. We create a new track in Logic naming it drummer, selecting input 1 and record him playing. 

Using the same microphone in the same input we get ready to record a singer. We pan the mic to the right and we've changed the bus from 1 to 2. We create a new track in logic selecting input 2 (making sure enable record is always checked when we want to record) and name it vocals. The singer has the headphones on and sings along to the drum track while we record it in to it's own channel. 

Or instead of panning the track what we could do is using the mic in the same no 1 input on the snake is to connect a cable from patch bay input 1 from the wallbox to out put 2 to mixer channel 2 on the desk. We could then select 3-4 pan left on channel 2 and the signal would be sent to bus 3 and to the AD/DA interface 3. We would have to create a track with the input number 3 corresponding with the AD/DA interface. 

Week 1 - Diary

Week 1 Diary Entry

I've found my first week as a mature student studying Music Technology entralling. I have already picked up a lot of new information that I didn't know and have gained some hands on experience with various pieces of equipment. 

I'm excited at the prospect of learning a lot through out the course and have already learned things that I can apply to my own musical endevours. 

The class seems to be comprised of a variety of young people who have different tastes and interest which will be very beneficial to my learning.

I feel very prepared as previous to the course starting I had started reading various books on recording, mixing and sound engineering techniques and having guided learning is helping to "lock" this information together in my head. 

The tutors we have are excellent and seem to be working in conjuction to give us a good understanding of the subject without them repeating what the others have said. 

Malcom mentioned he hopes that we will leave the college praising the tutors which will attract other students, well in my case this has already happened. Many of the successful musicians I have met and worked with have been through this course and all speak highly of the tutors. 

I'm hoping to maintain a high level of discilpline through out the course and am aiming for distinction grades. 

Overall the college is immense imho. The facilities are top quality, the resources comprehensive and the staff seem to really care about us working hard and achieving out goals. 

I'm concerned about my ability to use Mac computers and will try and see if the lrc can give me a brush up on using one, and having only used ableton since the start of the year it's going to be a challenge to learn other daws such as logic, reason, pro tools and the like. Although these are all multitrack recorders in essence the way each one works is quite different so it's a challenge I'm looking forward to taking on. 

It's going to be great to get experience an qualifications in an area I really love and enjoy and the way all the tutors are focused towards working in the industry and making money is reassuring as I wish to persue a career in the music industry.

Keyboard Skills and Sequencing

Keyboard Skills and Sequencing

Tuesday Lesson 1 4/9/2012

Today we completed a questionaire using the glossery on the vle about midi, eq, effects and various other audio related subjects. This was completed and uploaded via the vle therefore I don't have any notes to update about this. 

Sound Creation and Manipualtion, Acoustics and the Music Industry

Sound Creation and Manipualtion, Acoustics and the Music Industry 

Tuesday Lesson 1 and 2 - 4/9/12

These two lessons where an introduction into some of the topics we will be covering over the course of the year. 

We discussed how knowledge is power and by understanding the protocalls (the rules and regualations/instructions) will enable us to work well in the music industry. We need to know the rules inoder to adhere to them correctly. 

The parameters of sound are volume (Decibels or dB) how loud or quiet it is. The pitch which is the frequency of a sound (is it high or low).

The threshold of human perception of sound is about 20hz - 20,000hz. It's only "about" as adults can't hear the entire range. Radio singles and wifi are sounds that are present nearly everywhere yet are outside the hearing of humans otherwise we would hear wifi and radio all the time. 

Sound travels at about 340 m/s in air. A train travelling towards a train station you would hear the sound coming through the steel railway as sound travels faster the denser the medium. Sound would travel at about 5,500 m/s through the steel rails. 

This can also be demonstrated in a bath tub. If you were listening to music in the bath and submerged your head the sound signal would sound speeded up as water is denser then air. 

We also talked about health and safety. The number 1 cause of hearing damage is caused by earphone buds. These aren't allowed to be used in the classroom and can cause damage as the sound is going straight down the ear channel and into the tympanic membrane (which is a thin layer of skin in the ear).

When using over ear head phones there is a gap of air between the ear canal and speaker and air is spongy and provides some protection from the sound. This can be demonstrated with a balloon or airbed as when filled with air they are squishy. The bones inside the ear are the smallest in the body.

 The no 2 cause of hearing damage are cottom buds, and the no 3 cause is the workplace. 

Another big issues is RSI. Repetetive Strain Injuruies are caused when your doing repetetive tasks. The number 1 cause of this is videogames, and using a keyboard is no 2.  

We also started looking at other health and safety issues. As we are at college there are students under the age of 18 and a risk assesment will have to be carried out to assess any risks in the venue. This sometimes called "recky" or reconisonce. 

The no 1 concern is electricity and knowing where the breaker in the venue is can be extremely important.

We started talking about equipment theres 3 different types. 

Consumer

Semi - pro

Pro

Consumer products are for the average person and generally break down with 3 years. 

Semi pro products are of a higher quality and less likely to breakdown

Pro equipment is top of the range and should last a lifetime. It's easy to get spare parts for and repairable. Such as the Roland Tr707.

Sound on Sound is considered a pro level magazine. They review equipment on an unbiased nature and report on the pros, cons and improvements of various equipment. 

We started talking about a few different assignments we will be covering in future such as speech edits (changing the wording from "I really do not like the conservative party" to "I really -- Like the conservative party"). This is sample editing. 

We discussed a little DSP which is Digital sound processing. 

The first assingment we will be doing will involve creating four different wave forms, as some are shallow and sharp others are jagged rough and distorted to create a 16 bar loop where at one point all the wave forms will be played at once. 

We will be tasked with creating specific sounds and have to create them from scratch. Such as a bright bell with sharp attack with very little reverb. 

Reason

For the final hour of the second lesson we started to look at Propeller heads reason. We opened the program loaded up a Subtractor synth looked at the backend wiring, created a second subtractor synth and added a reverb unit. We added a matrix which enabled us to play back sounds and had a fiddle with synth to make various sounds. 

Evidence

We also talked about various ways of presenting our work such as a blog, video evidence, audio evidence, and screen shots. It will be important for us to use multimedia evidence to support our assignments so we can show our understanding of what we have learnt. 

Live Sound

Live Sound

Monday Lesson 2 - 3/9/12

Mixers, Mics and Foldback Monitors

The lesson was held in the theatre where we were shown a few things about microphones, mixers, monitors and health and safety. 

We started by collecting some equipment from the area above the main theatre. We got 2 foldback speakers and a mixer. 

We ran a cable from a power supply through some surge protected power bricks. It's important to use surge protected power supplys to protect the equipment and also because if there was a fire it wouldn't be covered by insurance.You would usually duct tape the cables down for health and safety to prevent people tripping over the cables.

We use IEC power cables (also know as a kettle lead) to provide power to the mixer and fx unit. 

We discussed two different microphones the shure SM58 and the SM58 Beta A. 

The Sm58 is the current industry standard. It is a dynamic mic designed to take a beating and is often seen at venues, on tv and is one of the most commonly used mics. It's priced around 60 pounds it captures low and mid ranges and can be used for vocals. 

The Sm58 Beta A is also a dynamic mic with a few differences. It captures some high end and has a feedback response unit which provides less feedback while monitoring. It's priced at 120 pounds and is starting to become the standard. The price being double that of an sm58 is one of the reasons the sm58 is the industry standard at the moment. The Sm58 beta A is great for capturing vocals where a good performance on an Sm58 would sound great on the Sm58 beta A.

We connected the mixer to an IEC cable to the power supply and also connected the active foldback speaker to a power supply as well. Making sure the volumes are down on the mixer, we connected a (mono?) jack cable from the active speakers output to the passive speakers input and connected a (stereo?) from the jack output on the mixing desk to the jack input on the active speakers balanced line input. 

A mono cable is indentified by a single black ring around the plug and a stereo cable is identified by two black rings. 

The monitors are magnetically shielded to prevent damage to computer monitors. If it was not shielded this could destroy a computer screen. 

We then turn the gain on the first channel (which is where the microphone is plugged in) to 12 o'clock maybe a little bit more, increase the fader on channel 1 to send the signal to the master output and push the fader on the master up. We also turn up the volume on the monitors so we can hear whats coming through. Hearing the sound isn't massively important at this stage as we will see the led volume meters moving if there is a sound coming through. 

We check the level of the microphone (either 1-2, 1-2-3 or by telling a joke!). We adjust the volume to make sure the at the loudest use the meter won't go into the red as this would cause distortion. 

Mixers have different amounts of channels, the one we was using had 8 mono channels and 2 stereo channels.

Now that we had connected up a microphone we talked a little about eq. Eq stands for equalisation. Eq is used for several things one is adjusting the sound coming through the microphone. If the vocal was coming through very boomy we could turn down the low and get a less boomie sound (or increase for a big boomie sound!) and we can adjust the high end to get a brighter clearer sound by turning it up a bit it can make vocals sound nicer and clearer. 

We then looked at adding an effects unit into the mixer. We turn down the faders so that we don't have any loud bangs or pops while connecting audio equipment We then removed the connection from the speakers and put a jack from the mixers channel out into the effects unit in, and connected a jack cable from the effects unit output to the mixers auxilary return. 

We removed the microphone from the xlr input and connected a mono cable from a guitar into the first mono channel on the mixer. As the guitar is a line level instrument the signal will be very weak we could mic up an amp and connect this to the xlr input,. but these days it's much more convienient to use a DI box between the guitar and channel input to boost the signal. 

We didn't use a di box for this instance and the signal came through very quiet. We was going to look at some effects on the fx unit but there was a problem and we was unable to do much more.

Health And Safety

Health and safety is massively important in the music industry. Working with artists, venues and the general public it's important to have a grasp on the various risks involved in live and recording environments.

Cables - All cables should be duct to the floor to prevent anyone tripping up.

Surge protectors - protect equipment and can prevent fires. Microphones/speakers can be blown if theres a surge.

Heavy equipment - can be dropped on feet or people damaging both equipment and the person involved. 

Pop/loud noises/feedback - can damage equipment and ears. 

Liquids/Food - Can damage equipment such as mixers and cause fires. 

Overcrowding - Can cause problems such as crushing or if theres a fire risk to getting out safely. 

Mic stands - Can fall over damaging equipment or hitting someone.

Mixing desk/movable equipment - If breaks aren't set on movable equipment this can be a hazard to both the equipment and people. 

Safety breakers - Knowing where the breakers are at a venue can be vitally important if theres a fire or electrical accident.

Cables - If their not correctly coiled or duct to the floor can be tripped over.

Running - Can be dangerous you could fall over and hurt yourself, or fall into equipment and damage the equipment.

Checking a monitor with ear next to it - can be very dangerous can seriously damage hearing if someone turns volume up or shouts into a microphone.

Sound Production and Music Recording

Sound Production Skills / Music Recording

Lesson 1 - Monday 3/9/12

Signal Flow

Today we started to have a look at signal flow. Signal flow is how the sound travels through the equipment. It's important to understand signal flow so you can identify problems (and solve them) and also so that you can change where the sound is going.

Firstly the vocalist sings into the mircophone. The microphone converts the acoustic sound energy into electrical energy that represents sound.

We use a XLR (grounded left right) balanced lead (as a balanced cable enables hums and buzzes the lead picks up to be removed) to connect the microphone. If we used an unbalanced rca cable that would pick up interference.

In the college this then goes into a snake box which connects to a wall box that takes the signal into the control room. Not every recording studio has a snake boxs but all will have a wall box. The snake box is numbered and this matches up with the patch bay and mixer.

The sound then goes into the patch bay which is wired in to the corresponding channels on the mixer.

A Stereo out from the mixer will go into an amp and speakers for monitoring.

The mixer is also connected to an audio interface AD/DA which converts electricity into binary so that the information can be processed by a computer (mac or pc) running music sequencing software. The Box can emulate instruments, equipment and is also used for mixing.

After discussing signal flow as a group of 6 we attempted to connect up a microphone to see if we could get a signal going through from the live room to the control room.

We set up a stand with microphone connected an xlr cable between the microphone and snake box, then went to the control room where logic was loaded on the pc.

We came across several problems while trying to set this up. We managed to get a microphone sound to come through the mixer and the speakers but there was a problem getting the signal into the computer and Dan Morgan had to come and have a look. It was apparent that there is usually some sort of problems happen usually occur and that sometimes these are not fixable in a short amount of time. This can lead to recording sessions having to be abandoned.

Setting up a Microphone Stand Correctly  

We was then shown how to set up a microphone stand and asked to set one up correctly.

A Microphone stand consists of the base and the microphone boon (holder). The first step is to set up the base. There is a knob which releases the central pole and allows it to be raised. It is important that the central pole be extended all the way out so that the tripod feet can rest flat on the ground. This is important because sound can travel along the ground and up the metal shaft and cause interference. The tripod feet have rubber stoppers to prevent this but these also stop bass sound escaping to the floor which would cause a loss of signal. With the feet folded out and the central pole fully extended we adjust the mic boon. There is another knob on the boon which allows us to raise or the lower the mic boon. This is loosened and is used to adjust the angle of the boon. The boon is adjusted and rotated so that it extends and is in line with one of the feet, this makes sure that the microphone is more stable and less likely to fall over and cause an accident. 

We then attach a microphone clip onto the end of the boon and it's time to set up the cable.

First we slot the xlr female side through the mic clip and (unlike the photo!) wrap the cable around the microphone boon. We then wrap the cable around the main pole of the stand and make a small coil of cable at the stand base and tuck it a little underneath it. This is in case we need to move the stand in future and to prevent accidents. We then clip the cable to the stand with a cable grip and we are ready to attach the microphone. 

The microphone slots into the clip and the female end of the xlr cable is connected to the male socket on the mic and we have a correctly set up microphone stand.