Showing posts with label unit 2. Show all posts
Showing posts with label unit 2. Show all posts

Sunday, 16 June 2013

Glossary

ADC
analog to digital converter works by converting a real life analog sound into a digital sound by converting the sound waves into binary code numbers. It does this by using  digital-to-analog converter however, since sound waves can only be picked up using the digits 0 and 1 which works as on and off switches. Here is a link to a video explaining this-





Alias frequency-



Binary

Because we as a people aren’t programmed to work thingsout in binary code, we work out things in something known as base 10. Forexample when we write something like the number 23,469- we’re doing this byusing the 10’s 100’s and thousands column. The only way that computers can understand something is by interpretingit into binary 2- which means that the only way a computer can understandsomething is using the digits 0 and 1. When using something like binary, the onlyy digits that can be used are: 1, 2, 4, 8, 16, 32, and 64.  For example, making something like the number 30,we would have to use this If we were trying to get to the sum of 3.0. In binarythis would look like: 11110




Bit depth ‘Bites’- this is just information in the form of 0’sand 1’s, so that computers can understand them, which can then be recorded for eachsample. There are thing called ‘bit depth’ for example you could have say ‘4bits’ or ’16 bits’ of information but the largest of all is ‘24 bits’ (6x4Thelargest bit depth is 24 bits because we would not be able to pick up anythingmore than that, as we are human.
Clipping- The term ‘clipping’ is used when something isincredibly loud. Something like a amplifier has a limited range of volt output,if you try and stress it, then it will sound distorted, and there is only a certainlevel of voltage output.  Let’s say wehad a diagram explaining this. The highest frequency would be ’10 DB’, whichmeans that anything louder than this will make it sound ‘piece’ and distorted.


 DAC- A digital audio converter takes digital info andconverts into audio info. This, as a whole, works as an entire system.  For example, I can take something like acomputer or even an mp3, and in turn, it will take the digital information andchange it into audio information, and it does this by changing into 0’s and 1’s.You can then get something like a stereo or headphones and listen to theplayback.


 DBFS- We use ‘Decibels’ to estimate something known as ‘soundpressure’. Whenever a decibel reaches its peak we call that ‘full scale’.Although it’s unexpected, 0 is the highest level of decibel- (full scale) butwhen turned down, it goes down into a minus value (-1,-2,-3). 6dBFS is the maxlevel below minus that a dB can hit (-6db).


Dither- Dither is an incredibly faint, and very low-It givesoff more of a white noise effect. For example when changing something that’s 24bits, and you’d like to change it to 8 bits, so that the data can fit ontosomething like an mp3 or a Cd then you can fit in the dither effect so that youdigital signals can run without distortion. There is no way for us topick up ‘dither’ in our ears. All dither does when used, is allow the tracksusing dither to run clearer and cleaner- making it sound normal/natural.




Digital audio interface- digital interface is what makes the connection between the 



 Digital effectsprocessor-



Digital mixing console-



Sample rate- sample rate is the amount of analog data record in 


Understanding Technology

Question 1: What is equalisation, and how has it changed the working process since the early days 
of studio recording and mixing?

Answer: EQ (equalisation) is something that every sound has. to understand EQ you have to know what Hz is, Hz is usually measured by how much the sound vibrates per second. The way EQ works is controlling what amount of Hz works and i am going to give a quick example.
having the EQ at 2000Hz and then changing the dB to +5 dB effect or increase the high frequency of the EQ, we often use EQ to control the sound of things like snare drums, kick drums, bass, guitars, vocals etc. 
the reason why it is so vital to have a microphone that picks up good equalisation meaning low and high frequency is because it gives you a clear sound. 
But having a microphone that picks up too much can sometimes not be helpful because sometimes we want to clear some extra noise like: extra breathing on the vocals, or moving about noises that you get during recording. Before people like 'Frank Sinatra' had to use U47 Valve Capacitor microphone which only picked up a certain amount of EQ which could not be changed and if they wanted the EQ to change they had to look for different microphones that had the amount of EQ that they wanted.
Once EQ started being used as a mixer people did not have to change microphones as much because the pioneer's made channels. Channels works like this: you have the dB which controls the gain on the channel and then you have a separate panel that controls the high frequency, mid high frequency, mid frequency, mid low frequency and low frequency, the way this works whilst recording a band is that if you used a microphone on a snare you would maybe eliminate all the low and mid low frequency and  increase a bit of the mid high frequency to get that high pitched sound to the drums but also have a bit of that mid range sound that you need on the snare. 

Question 2: What is the polar pattern called that the U47 used for recording vocals? What is the polar 
pattern of the other microphone that was used previously? Why do you think the 
directional ability of the U47 helped in the recording process in this particular case? 

Answer:

Part A: the polar pattern for the U47 is cardioid and figure of eight it has an extra polar pattern more then the ribbon microphone which makes it more reliable for more different parts of recording then the ribbon microphone.

Part B: the polar pattern for the ribbon microphone which was the microphone that they used before the U47 is the figure of eight polar pattern.

Part C:The ribbon microphone has a figure of eight polar pattern which the U47 microphone  has, but the U47 has an extra cardioid polar pattern which is amazing for recording vocals, the reason for this is because it stops the sound from the opposite direction affecting the sound that were trying to make with the vocals, it also has a more focus range which is great for avoiding feedback from the speakers during live recording.

Question 3: What is meant by the term 24/48? Why do you think, in this case, using 96k was not suitable? 

Answer: 24/48 is a sampling rate that picks up on the cd 24000 or 48000 Hz, explaining in more detail the sampling rate is how much data recorded on to the CD, having a 96k sampling rate is great for modern recording because it is most used for DVD'S and current high definition recording, but when using 96k on a older recording it picks up a lot of unnecessary data which could end up having a major effect on the final recording. 

Question 4: What is the job of the ‘converters’? How important are these in a modern studio?

Answer: the job of a converter is to convert real sound (analog sound) into digital sound. The reason why this is so important in a modern studio is because we use a lot of digital but we use the same amount of analog and being able to convert analog to digital helps us have a wide range of real life recordings, also having analog recording converted to digital helps you have more control of the analog recording once it has been converted to digital as in you can add more treble, decrease the bass, compress the sound and all sorts of effects etc. 

Question 5: What other benefits do using DI boxes give an engineer in the studio?

Answer: what a DI box is also useful for is home studios, in a lot of home studios space can be a real problem because of the amount of gear you need. So having a DI box means that you do not necessarily need microphones meaning you have a lot of room for other things and you do not need to waste room on microphone stands. It is also useful to get a different sound from a microphone recording so if you record an instrument through a DI box and a microphone at the same time, you might get a more realistic recording with real reverb recorded from the microphone. 

Question 6: What would you suggest?

Answer: my advise is to have microphones for all the vocals, guitar and bass running through the PA system but not for the drums because the drums is naturally loud, but if you want to record a stereo mix onto reaper i would suggest to record the drums first separately, but my guess is that you want a good sound recorded but live, so i suggest instead of plugging everything to the PA and playing it out loud, get headphones for everyone that is playing and plug it in to the Alesis MultiMix 8 so each person can hear themselves and not get feedback.  Another thing i would suggest is to use the mixer to record the vocals, bass, guitar and have one microphone recording the drums from overhead which wont give you a lot of control on the sound of the drums but it will get you to record everything without feedback. 

Friday, 31 May 2013

Studio Assignment

Question 1:
+4dbu and -10dbv are two measurement units that describes the signal level. The difference between the two is that +4dbu is a measurement used for balance line level connection. A balance line level connection is often used for professional studios, examples of equipment that we use that has balance line level connection is a mixer, a guitar amp, a keyboard, a interface and many more specific studio equipment. A balance line level connection can only be used for equipment that takes on +4dbu signal levels, the reason for this is instruments like amplifiers or mixers uses a lot of faults and if you used a unbalanced cableing connection there wouldn't be enough power to supply the amp or the mixer. A unbalanced connection measurement unit is -10dbv, this means that unlike a balance connection a unbalanced connection is weaker it uses lesses power and if you was to supply a mixer with the un balance connection it would litteraly distort the whole sound due to the amount of energy being used. Things that uses a balance connection is a mp3 player, a mobile phone, a laptop and computers. 

Question 2:

A cable that you would use to plug in a mp3 player or a ipod to connect into a mixer is a rca phono connecter cable, this cable has a central tip and a sleeve, they are unbalanced and any equipment like a cd player or ipod or iphone. A phono jack has one contact point that makes it unbalanced, but you can plug it into a mixer and get a unbalanced equipment eg. Phone to connect with a mixer which works. 

Question 3: Mic A because Mic A gives a wide rage of frequency both low and high frequency
it

Question 4:

20Hz - 20kHz,


Question 5:

24 bit wordlength, 44.1, 48, 88.2 sampling rate.

Question 6:

The most important piece of equipment that the most amount of money shoud go to is the audio interface. Having a good audio interface helps the quality of the sound recorded from a mic or a guitar and it also helps the quality of the sound on the monitors. 

Question 7:

A specific thing that people should not do with monitors is play it out on full blast, they should play it at a level that does not do long term damage to the ear. Also they should put the monitors at a place that's high but not to high, they should also make sure they know what type of line signal level the monitor is to prevent any serious damage to the monitors.

Question 8:

First thing I would check is the equipment that they have such as; the amount of xlr cables, the types of amps, amount of jack cables, also what microphones uses phantom power, and also what type of microphone they have. After all of that I would check the interface and I would check what type of interface it is and what type of mixer they have.  

Question 9:


Supply a list of all equipment, with explanations, pricing and wiring configuration (block diagram), that you would recommend for the following situations:


balanced level signal equipment
KRK RP-5G2 RP5 Rokit Powered 5 Studio Monitors
price £249,99



M-Audio M-Track - Two Channel USB Audio/Midi Interface

£79.00



Mic stand x 2
£9;99


Midi - Keyboard:

Avid KeyStudio 49-Key USB MIDI Keyboard + Pro Tools SE - £79.99


Condensor Microphone:

Audio Technica AT2020 Cardiod Condensor Studio USB Microphone -  £88.85

Headphones:

Noontec Zoro Studio Professional Headphones RED HD On Ear Genuine High Quality - £41.59
part 2
microphone stand - £7.65 x5 = £38.25
John Hornby Skewes Herald 5 Piece Drum Kit.- £249.99
suzuki digital grand piano - £1.599
suzuki grand piano includes:
*three dimensional stereo imaging sound
*80 watts of power
Digital Mini-Grande Piano
Matching Bench
Demonstration SD card
USB SD Card Reader
Student Series Tuition Course (Book 1)
Instruction manual
AC cable
Assembly Guide
Tools for Easy Assembly


Guitar - Lindo SLIM Red Electro/Electric Acoustic Guitar+FREE Carry Bag and Accessories -

 £167.49

sE Electronics SE2000 MKII Mic + Reflexion Filter Recording Bundle - £169



sE Electronics SE2000 MKII Condenser Microphone:

Cardioid polar pattern

1" Gold-sputtered capsule with mylar diaphragm
Frequency response of 20Hz-20KHz
125 dB Maximum SPL for THD < 0.5
Sensitivity of >16m V/Pa -36*+/-2dB
48V Phantom power necessary
Output impedance of < 200 Ohms
Output noise of < 17dB typical (A weighted)
Rubberised noise-dampening mic clip (included)

Boom Mic Stand by Gear4music:


Mic Thread: 3/8" (10mm)

Min Stand Height: 35" (90cm) (floor to boom swivel)
Max Stand Height: 64" (163cm) (floor to boom swivel)
Folded Length: 36" (93cm)

XLR (F) - XLR (M) Microphone Cable, 6m:


Length: 6m / 20ft

Jacket and Insulation: PVC
Spiral Shield: 64 x 0.12mm
Conductors: 20 x 0.12mm
Connectors: Female XLR to Male XLR
Colour: Black

Microphone Pop Filter Shield for Mic Stand:


Stops damaging noises from 'p' and 'b' sounds

Protects your mic
Flexible gooseneck holder
Swivel mount for easy positioning
6" diameter circular pop filter
Universal mic stand clamp




Features/Specifications:

  • Ultra-compact headphone amplifier system for studio and stage applications
  • 4 independent stereo high-power amplifier sections
  • Highest audio quality with virtually all types of headphones even at maximum volume
  • Phones Level control plus accurate 6-segment LED output meter per channel for easy level monitoring
  • 2 balanced stereo main inputs with independent Level and Balance controls for individual mixes, selectable for all 4 headphone amplifiers
  • Accurate 6-segment LED main level meter for precise level indication of both line inputs
  • 2 headphone outputs per channel (front and rear panel) allow a total of 8 headphones to be simultaneously connected
  • Parallel Link Output allows cascading of several headphone amplifiers
  • High-quality potentiometers and illuminated switches for long-term reliability
  • All Mini Series models can be stacked on top of each other to create an ultra-compact signal processor solution
  • High-quality components and exceptionally rugged construction ensure long life