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Daisy Grace Wakefield 22 October 2015 Better Mixes Page 1

Unit 2 Write Up PDF

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Page 1: Unit 2 Write Up PDF

Daisy Grace Wakefield 22 October 2015

Better Mixes

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Daisy Grace Wakefield 22 October 2015

T4.2

Static Mix A static mix is one which uses panning but does not use any

automation.

Out of the three mixes i created, automation was used on all mixes and therefore i did not create a static mix. Reasons for this include;

• On mix 1, within the football stadium, once the performers had sung their football chant, they immediately started talking and creating irrelevant material and therefore, using automation to slowly fade them out was much more effective than just cutting out the undesired audio. The fade tool was used in conjunction with this, but automation was needed towards the end to help the audio fade out faster.

• For mix 2, Brotherhood war scene, it would have been a static mix if the original audio was desired for the majority of the piece,

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beginning and middle, and faded out just before the end to have an effect on the audience. As automation was used to achieve this, it cannot be a static mix.

• For mix three, Africa, automation was used to bring in the guitars during the chorus as even with the use of compression and EQ, the guitars took over the mix and other instruments such as vocals could not be heard.

Signal to Noise Ratio (SNR) SNR is a measurement which measures the level of desired

signal to that of unwanted signal or background noise. Sometimes background noise is desirable, for example, if we were to mic up a drum kit in a church, we would not place the microphones close to the drums as possible like we would in a studio. There is a reason that the drums are being recorded in a church and therefore we desire the echo effect that the location will give when playing the drums. Other times, the background noise is not desired as things such as breathing and the noise from the air conditioning can be picked up and this is what we do not want.

In my first mix, using EQ to remove unwanted frequencies of specific instruments but cutting away them away, meant that any unwanted background noise would have been removed from the mix. In this mix, the alto saxophone kept picking up the kick drum, due to their locations when recording was taking place. By cutting the active

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Automation

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frequency of the kick drum, roughly 30Hz to 200 Hz , this removed the unwanted noise. This allowed me to

carry on with the mix as the kick drum would have been overpowering.

Using automation is another effective way to remove unwanted noise. Examples of background noise or disturbances could be heavy breathing, scuffling of paper, squeaking chairs. These noises can be removed this way without removing too much from the mix like if you were to cut frequencies using EQ.

It would have been better if the instruments were recorded individually as this would reduce the amount of unwanted background noise and bleeding from other instruments.

dBFS and Headroom dBFS stands for decibels relative to full scale in the digital world.

In a digital system, the highest value we can have without any distortion or clipping occurring is 0 dBFS. As 0 dBFS is the highest value in digital terms, all other numbers will be negative.

Headroom is the space between the signal level being received and the maximum level the signal can be without distortion or clipping occurring (0dBFS). Looking to the faders on logic, you want the recording levels to stay in the ‘yellow zone’ which is roughly in-between -6dBFS and -12dBFS, which is classed a having good

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headroom on a mix. This also helps to avoid exceeding, or peaking above, 0dBFS, distortion and clipping as these in the digital world are not desirable.

To avoid very little headroom or no headroom at all within my second mix, i found it was important to make sure that the levels being received through the pre-amps, focus rite and apollo, while the performers were practicing their piece, did not exceed 0dB on Logic’s faders. I used the gain

control on the apollo (vocals and keyboard) and the focus rite (drums) to alter the signal levels. This would have caused problems for me when i started mixing, as this would mean that it would have been harder to create a mix without potential distortion.

Dynamic Range The dynamic range is the difference in decibels (dB) between the

quietest and loudest point of the waveform.

Different genres of music have very different dynamic ranges for example, if an orchestra was playing a classical piece, it will have

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Headroom

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obvious quiet and loud sections to the piece as they would use the acoustics of the room they're in to create their unique sound, but with pop music, this level is

set at the same level throughout with very small dynamic changes. Compression can alter the dynamic range.

For mix 1, the sine wave shows that there isn't any digital distortion and that there is a good dynamic range of the audio file.

Also for mix 2, there isn’t any distortion. It

does seem to peak at the highest point on the audio file but there is a small gap between the

highest point and where distortion would occur.

For mix 3, its clear that there is a very high level of distortion occurring within

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Image 1.1

Distortion

Highest points

Lowest Points

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the mix. Looking at the sine wave, it makes it easier to understand why digital distortion is avoided as there will now be a very noticeable fuzzy, raspy sound.

EQ / Frequency Spectrum EQ can be parametric or semi-parametric. Parametric EQ means

that when we EQ an instrument, we are able to control the width, decibels and hertz being cut or boosted and semi-parametirc EQ means we can only manipulate decibels and hertz.

When we use EQ, we listen to the instrument selected and cut away at the frequencies not associated with that particular instrument. Frequencies associated with each individual instrument are called its ‘active frequency range’. Different instruments have a different active frequency range meaning that between certain hertz, the instrument will produce a signal.

For example, a Pianos active frequency range starts at 30Hz and going up to 4KHz, therefore we would cut below 30Hz and above 4KHz. Cutting unwanted frequencies allows more headroom within your mix as this can get rid of any unwanted background noise (SNR).

Once unwanted frequencies have been cut away from instruments, we can look at ‘boosting’ frequencies. This means that when listening to the track, if there are any instruments struggling to be heard, we can boost them by raising the volume of that specific instrument dB

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at a time. By using a paramedic EQ tool, we are able to select the frequency range, width, that is too quiet and increase its volume, dB.

Within my second mix, i used EQ to cut frequencies between 5kHz and 20KHz of the kick drum and boosted the lower frequencies of the

snare, left over head and right overhead microphones as the cymbals and hi-hats

were very strong in the mix and other parts of the

drum kit were buried.

As all instruments were playing together in the same room, there was always going to be bleeding of instruments and by cutting away at the dominant frequencies and boosting those lost in the mix after cutting, this allowed instruments hidden to be heard.

For my first and second mixes, as the instruments were recorded in the same room at the same time, the drums were miked up with one kick drum microphone and overhead microphones in a spaced pair manner. Despite only 4 microphones for the drums, they still bled into others which meant carefully selecting the right frequencies of other instruments to cut or boost to create a good balance within the mix.

Digital Vs Analogue Distortion An analogue signal is a signal that does not contain breaks, in

other words is continuous. For example, our hearing is analogue as we are hearing in real time, as the sounds reach us, and there are no breaks, and analogue signal varies in amplitude and frequency.

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Kick, Snare, Left overhead, Right overhead

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A digital signal contains breaks and can only have two values, 1 (on) and 0 (off). A digital signal is sent in a series of pulses containing ones and zeros.

Digital distortion can result in a raspy and fuzzy sound. Digital distortion occurs when the signals shape is altered in some way. This can occur when a system such as an amplifier cannot cope with the full signal and so cuts come of it off called clipping. This results in a change within the dynamic range. Software will have some kind of indictor which tells you its dBFS limit and as to where clipping starts to happen.

When talking about analogue distortion, this is ‘referring to the character that the analogue processing/recording equipment and the recording medium add to the sound’ [1]. http://www.soundonsound.com/sos/feb10/articles/analoguewarmth.htm

With my first two mixes, during the recording session, i made sure that there wasn't any digital distortion by adjusting the gain levels on the pre amp before recording making sure that we did not peak

above 0dBFS and the levels were in-between -6dBFS and -12dBFS, therefore leaving enough headroom.

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Enough room so no distortion

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Looking at the mixes sine wave it is clear that no distortion took place.

As for my third mix, Africa, looking at the sine wave, it is obvious that so much distortion is taking place. Avoiding this is key to a good mix.

Compression

Threshold Compression is used to make loud sounds appear softer and can

reduce an audio files dynamic range.

Threshold can be called the ceiling as when the signal hits the threshold ceiling, this is the set point where automatic volume reduction starts. This means that everything above that signal becomes quieter. A threshold of -10 is much lower and will not have as much of an effect ton the audio signal as one of -40. Below the threshold, the compressor has no affect as the compressor only acts on signal on the other side of the threshold. The compressor reduces the volume to stop the input signal from getting louder and potentially avoiding digital distortion.

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For mix 2, i set the threshold on the left overhead microphone to -8.0dB and the right overhead to -12.0db. I did this so that the higher signals coming through from the cymbals and hi-hats, were not affected or lost within the mix. If i set the threshold on the cymbals and hi-hats to a high threshold, they would have been muffled and/or lost within the mix.

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Output / Makeup Gain A compressor will only adjust the volume when the signal

becomes too loud or gets too high. The ‘Make-up gain’ comes into effect and this lets you bring the signal back up to a healthy level and gives back the dB that was lost from compression. Adding make up gain will increase the dynamic range of your audio file. For example, once compression's taken affect, there will be a noticeable different in volume. If we say that -4dB has been lost, we can turn the whole signal up 4dB to make up for that loss, This will not affect the compression already added.

In mix 2, for the electric guitar, I added 1dB to compensate for the loss through compression. The signal received was already quite high and so i didn't need to increase the make up gain by that much to bring it to a healthy level.

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Ratio The ratio indicates how much the audio is compressed or ‘turned

down by’. For example if the ratio is 2:1, anything above the threshold is ‘turned down’. This means that if there is 2dB above the threshold, then only 1dB is allowed through. An alternative way of looking at it is that it has been turned down my 1dB, or 1dB gain reduction. 4dB above threshold then 2 dB are allowed through, 8dB above, only 4dB allowed and if 10dB the 5dB are allowed through. Essentially 2:1 ratio means half.

If we had a ratio of 3:1, every 3dB above the threshold we go, only 1dB is allowed through and so we will have 2dB gain reduction. If the ratio was 10:1, then the gain reduction would be 9dB, meaning that only 1dB is allowed through. If the compressor has a ratio of 1:1 then this means the compressor is doing nothing (neutral).

I set the ratio at 2:1:1 for the kick drum in mix 1. It took me a number of times to set, what i thought to be, a good ratio as i didn't want a large amount of volume to be lost.

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Attack and release The attack time is about how quickly the volume is reduced or

lowered when the input or signal exceeds, or goes over the threshold. If the attack time is too slow, then for a short time after this, you could hear bursts of loud, uncompressed, music coming through which would not be good for your mix.

The release time on compression is about how quickly the volume can return to normal when the input or audio is no longer above the threshold. If you allow the volume to come back too fast, then it will be obvious that the signal level is going up and down just as if you were turing the volume up and down on a CD player when listening to music or on a remote when watching television.

The use of a slower attack time, for mix 1, on the snare meant that you could hear the unique timbre of the snare and help add to the atmosphere of the piece as the accommodating video is set in a war zone, hence why the attack time is low at 10.0ms.

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For the release time, referring to the accompanying video, it was best to have a higher release time, of 68.0ms, to help add to the atmosphere of the piece as having a slow attack time and a low release time would cancel out the attack time resulting in a snare sound that is undesired and does not work for the piece.

Challenges within the mixing process

Challenges i faced within the mixing process included how to use compression correctly. I found it very hard to judge the threshold, ratio, attack and release as with compression, it is not only designed to make loud signal appear softer, but to also, if used correctly, to add energy into the instrument/mix.

I found compression to be a long task with numerous answers and faults. For example, finding out what to set the threshold to for each instrument, then the right/appropriate ratio, attack and release for it, i found myself going around in circles trying different ratios with thresholds until becoming satisfied with the outcome. I went searching for some kind of answer to ‘what do i set the threshold to’ but all came back with the same answer which was, there is no definitive answer but you have to trust your ears and listen.

The Cymbals and hi-hats in mix 2 i still feel are slightly too overpowering despite the use of EQ and compression on the kick drum, snare, left and right overhead microphones. I think that i should have spent more time trying to make them fit within the mix by experimenting longer with EQ and compression with the different instruments and microphones.

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I found it hard to get the electric guitars to sit right within my mix as i felt that despite playing around with automation, compression and EQ, they were still very dominant within the mix although their recording levels were good. I think it did not help that other instruments were clipping and the guitars had analogue distortion as i think this made them seem too loud.

How could you improve the mixes/ if i was to do this again.

If i was to do this again, i would have recorded more versions of the football chant for mix 1 as comparing the audio to the video, its obvious that the two don't work together effectively and to capture the scene effectively as one example of how it really didn't work was that in a football stadium when a game is playing, when people chant, they are not in time with each other and the audio used was too clean and in time to use effectively. More versions would have helped along with a lot of reverb added to the different vocal channels.

For my third mix, Africa, to avoid distortion occurring if i was to do this again, i would start off mixing the song with the use of EQ and compressor where needed and once i had done a fair amount to the mix, bounce the song and take a look at the sine wave to see if distortion is occurring. If distortion is occurring then i would go back and try again but if no distortion occurs then i would continue and repeat the bouncing process after i had progressed further into the mix.

For mixes 1 and 2, also three if i had been present for the recording process, i would have asked that each instrument was recorded separately, or that all DI (direct input) instruments are played in the same room, and checked their recording levels and used the

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gain on the pre amps, apollo and focus rite to set the right levels for recording by leaving enough headroom to avoid distortion and therefore making it easier to mix the songs as i would now have space to work with.

Bibliography

Image 1.1

From - http://www.planetanalog.com/document.asp?doc_id=527307

Acesses: 20th October 2015/ Copyright © 2013 UBM Tech, All rights reserved

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[1] http://www.soundonsound.com/sos/feb10/articles/analoguewarmth.htm - Accessed: 18th October 2015/ All contents

copyright © SOS Publications Group and/or its licensors, 1985-2015.

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All screenshots were taken from my work by me and all information is my own unless stated.

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