<![CDATA[Sack Records - Make A Record]]>Sat, 24 Feb 2018 02:02:39 -0800Weebly<![CDATA[Sound from Scratch]]>Wed, 03 Jun 2015 07:55:44 GMThttp://sackrecords.com/1/post/2015/06/sound-from-scratch.htmlEvery musical instrument has the ability to manipulate (to varying degrees)  the 3 properties of sound.
  • Pitch
  • Tone
  • Volume

The way in which each instrument does this is dependant on its physical design and structure. A guitar will always sound different to a trombone due to how each is constructed and how the sound is produced within them.

Every note played on any instrument is not just a single note, but a combination of strong and weak harmonics stretched across the audio spectrum. It is these differences in harmonic structure that defines the Timbre or Tone of an instrument.

Synthesisers are a good way to break down sound into simpler components so that we can get a grasp of how it all works at a fundamental level.

In this article we look look at the very basics of the components of a synthesiser and how they interact with sound at each point ultimately having the ability to completely hand craft and almost limitless variation of sounds.

The average synthesiser is made up of 5 stages. These are: 
  • Oscillator
  • Filter
  • Amplifier
  • LFO
  • Envelope

Where the initial sound is produced. The oscillator builds a user chosen waveform type (sine, sawtooth, square etc). 
These waveforms are a combination of fundamental and harmonic frequencies.
Each basic wave shape has its own tonal characteristics and is used as a starting point for the final sound that you are looking for. On its own however, without any filters or modulation these initial sounds tend to be harsh and unpleasant.

To begin shaping the sound a filter is used to cut out a portion of the frequencies, usually starting with the highs by means of a low pass filter.
The type and specific characteristics of the filter used will have a major impact on the overall tone of the sound produced.
One common technique is to create 'Resonance' by adding a slight bump at the cutoff frequency.

The Amplifier section simply does what amplifiers to best and act as a gain stage in the synthesiser chain.
The major difference between this and a standard amplifier however is that it is designed to be modulated in different ways, often very fast, to add sonic characteristics to the final sound produced.

The Low Frequency Oscillator is another point where a waveform is introduced.
Rather than being the basis to the sound being produced however, the LFO creates a shape for the signal so far to follow thus modulating a certain characteristic of the sound (ie. pitch) to create a sense of movement.

The Envelope instructs the sound how to behave once it has been engaged by means of a user controlled interface such as a keystroke.
In other words the enveloped controls how the sound changes over time from 'note on' to 'note off' states.
The Envelope control is almost always Attack (how long it takes the note to get to full volume), Decay (how long until the note settles back to its main volume), Sustain (the level that the note remains at until the note off instruction) and Release (how long it takes the volume to fall to 0 after being released.

<![CDATA[EQualise]]>Thu, 28 May 2015 01:06:30 GMThttp://sackrecords.com/1/post/2015/05/equalise.htmlThe term 'Equalisation' is a throwback to the days of needing filters to correct the tonal changes of audio sent along telephone wires.
These days EQ refers to any filter device that targets and manipulates specific frequencies within the audio spectrum. Picture
In this article we will come to terms with the basics of how to use EQ musically, and discuss a few pieces of 'Good Practice' advice.

  • Filter Types 
  • Cut vs Boost
  • Sweeping
  • Mirror EQ

Types of Filters
There are a few basic types of filters that you need to get your head around. Notice in the examples below that there is a lot of 'boosting' (increasing the level of a frequency) going on. These are exaggerated in order to demonstrate what they look like and aren't necessarily good EQ moves. Here are some rules of thumb that you can use as a guideline.

Cut Before Boost - The temptation is to boost first. This is because the effects are much more audibly noticeable and you feel like you are getting somewhere. In reality, every boost your make is adding more noise to your mixing and reducing your headroom. It takes discipline but know that cutting out frequencies in order to 'reveal 'the ones you want to stand out is better for your mix than simply boosting the ones you want to hear.

Use Your Ears - After a while you might start to think you know what a good vocal or snare drum looks like. Think about it for a second and you will see how ridiculous this is. Trust your ears over your eyes always. This is one reason some engineers prefer using outboard gear or plugins that don't have a visual representation of the EQ moves. People have been known to close their eyes or even turn their computer monitors of so they are not temped to look too hard.
High/Low Pass Filters

A Pass Filter (High or Low) allows frequencies beyond (above or below) a defined threshold past, while blocking or 'cutting' the rest.
Pictured is a High Pass filter as it is 'letting the high's passed'.

A High Pass Filter (HPF) is probably the most common of the filters and often can be engaged on gear at the source such as a condenser mic or preamp.
In effect, it takes out the mostly unusable low end that in most cases (barring kick drum and bass guitar) is only rumble and unwanted noise. Vigilant use of an HPF in recording and mixing can be a great head-start to a cleaner mix with more headroom.
Notice how it forms a smooth curve rather than an abrupt cliff. This is because in EQ smooth and gradual curves sound more natural that sharp 'notches'.

Shelf Filter
Next on the list is the Shelf Filter.
As demonstrated in the picture this type of filter engages at a set frequency then create's a plateau or 'shelf' for the remaining frequencies above or below the initial.
A shelf filter like this is the most practical for a natural sounding boost or cut, often used to add 'sparkle' or 'air' in the high end (similar to what is pictured).

Band Pass Filters
The final main filter type is the 'Band Pass' or 'Bell Curve' which can be used to cut or boost in a bell shape all across the mid-range. The size and shape of the 'Bell' is called the 'Q' and can be adjusted to suit your need.
The Band Pass filter is where you will get all your surgery (if needed) done, specifically notching out unpleasant frequencies.

A Clean Sweep
Sweeping is an extremely useful (if not essential) technique using a Band Pass filter. As shown above, you grab a narrow 'Q' band, exaggerate a boost, and slowly sweep back and forth, listening for any frequencies that are nasty sounding or muddying up your track. Once found you can 'notch' that frequency out (I recommend working in 3db chunks). Lots of little cuts like this might not sound super obvious in isolation but will add up to a much cleaner mix with more headroom.

Mirror EQ
Mirror EQ is a technique where you take two or more tracks and 'carve' them out their own space in the mix.
When you open a single EQ window and look at the background, you can think of that as your total amount of space for all the frequencies in your song. That means that all of your tracks need to share that same space. By making conscience decisions about which instruments should sit where and taking out frequencies of one track to allow another to use that space will go a long way toward having nice separation and minima; 'masking' in your overall mix

<![CDATA[Compression Baby!]]>Wed, 20 May 2015 03:19:12 GMThttp://sackrecords.com/1/post/2015/05/compression-baby.htmlAlongside EQ and Reverb, Compression is one of the 3 key tools in mixing (and recording and mastering) yet one of the most misused and misunderstood.

What a Compressor does is seemingly simple; It controls the Dynamic Range of a track by reducing the volume every time a peak passes beyond a set threshold, then 'releasing' the signal back to it's original level.
Its other main function is to alter the shape or envelope of a sound.

Understanding compression and using one with intentionality can significantly improve the clarity, consistency and punch of a mix by adding body, character and definition to a track.

In this topic we will cover

  • What is compression
  • How it's controlled 
  • What effect its parameters have on a sound
  • A starting point to getting the sound you want

What is Compression?

Compression was originally a hardware device invented to automatically keep levels and dynamic range under control when recording. 
It is a helpful analogy to imagine an engineer with his finger on a fader, pulling it down when the signal gets too loud, and bringing it back up when it's back down to desired levels.
Although it is now very accurately represented by software plug-ins, a lot of engineers still swear by outboard (analogue) compressors. 

How it's controlled.

Your average compressor has 5 main controls (ignoring the input/output controls which are pretty self explanatory).
These are Threshold, Ratio, Attack, Release and Make-up Gain.

Threshold is the point above which the signal will engage the compressor. Below this point the compressor is inactive.
Ratio is how much compression will take place once the threshold is crossed, expressed as a ratio (1:1 = no compression, 30:1 = extreme compression i.e. limiting)
Attack is the 'reaction time' of the imaginary engineer, how quickly they bring the fader down after the Threshold has been crossed.
Release is how quickly they bring the fader back up to normal levels after the signal has dropped back below the threshold.
Make-up Gain is the point after compression where you can bring the whole signal back up to pre-compressed levels.

Practical uses

The 1st practical use for compression is to even out the dynamics of a performance. This can either be done post-production through plugin's or outboard gear, or it can be done 'on-the-way-in' while tracking.

Vocalists for example (especially those inexperienced in the studio) may tend to move around while performing or switch from whispers to shouts, thus increasing the overall dynamic range of the track, not necessarily a good thing as the quieter words will tend to get lost in the mix and the louder phrases might 'poke out' unpleasantly.

As previously discussed, by reducing the peaks of the track, you are able to increase the overall volume. This has the effect of essentially bringing up the quiet parts in volume without increasing the louder parts.

The 2nd main use is to further shape the sound of a track.
This can be done with careful attention to the various settings available on the average compressor, namely Ratio, Attack & Release. 
Without going into the pages of detail and discussion on this topic I will outline some basic rules of thumb. As the effect of compression is most noticible when the source is transient in nature (a rapid spike in amplitude) e.g. a snare drum, I will use that as an example for the following.

Fast Attack, Fast Release

This setting 'catches' the transient almost immediately and then quickly releases it. On a snare drum this has the effect of reducing the impact of the hit, but sustaining the body and tail end (ring) of the sound.

Fast Attack, Longer Release
With this setting the initial 'hit' of the drum and the 'ring' or sustain are decreased by about the same amount. This is useful if you want to even out the performance without much changing the overall sound of the drum.

Slower Attack, Moderate Release
With a slower attack setting the initial transient of the sound is allowed to pass through. A setting like with like will emphasise the impact of the initial hit by reducing the ring and sustain of the note. This is useful for adding definition to drum hits or tightening up under dampened drums.

These examples are by no means everything a compressor can do but they are a good starting point to get your head around what effect difference settings have on the signals waveform as well as how these impact the overall sound of the track.

The Threshold and Ratio also extremely important but as their settings are entirely dependant on the specific signal you are working with, we can't really outline any rules of thumb. Other than to say that you set the Ratio based on how heavily you want a signal to be compressed, and set the Threshold by first raising it above the signal so that no compression is taking place, and steadily reduce it until you get the desired amount of Gain Reduction.

<![CDATA[Tools of the Trade]]>Thu, 14 May 2015 02:58:26 GMThttp://sackrecords.com/1/post/2015/05/tools-of-the-trade.htmlWith an unimaginable quantity and scope of plugins/hardware available it is all too easy to be overwhelmed and/or fooled into thinking your mixes can never sound professional without a serious investment in this gear.

There are in fact 3 tools that you need to get great mixes and all of them come free with your DAW.
As with any tools, knowing how to use them is key.
In this post we will look at the 3 categories of effects and the basics of what they are used for:

  • Filter     -  Related to the Timbre (Tone) of the Signal. 
  • Dynamic - Related to the Amplitude (Volume) Volume of the Signal.
  • Delay     - Related to the Propagation (Space) of the Signal.


Careful and intentional use of these 3 elements is key to getting professional sounding mixes.

Second to capturing great recordings at their source, these tools are at the heart and soul of mixing and they all come free with your DAW.

In this article we will outline the 3 main effect categories and give a brief introduction to how they are used.

Filter Effects
  • EQ
  • High Pass Filter
  • Low Pass Filter
  • Wah

Filter effects have the ability to independently alter specific frequencies of an audio signal.

This is most effectively used through EQ and High/Low Pass Filters to remove unwanted frequencies from a track.

Dynamic Effects
  • Compressors
  • Limiters
  • Expanders
  • Noise Gates

Dynamic effects impact the amplitude of the audio signal and are widely used in audio production.

Dynamic effects can greatly enhance your mix but only when used musically and intentionally.
Understanding how they work and how they affect they have on the sound is crucial.

Delay (Time Based) Effects
  • Reverb
  • Delay
  • Echo

Delay effects alter the timing of a signal and are often used re-create the sound of an acoustic space in post production.

 They can be used to 'glue' different instruments together by using a common reverb across multiple instruments thus giving the effect they that were recorded in the same acoustic space.
<![CDATA[Get Ready to Record Into Your DAW]]>Tue, 05 May 2015 04:35:16 GMThttp://sackrecords.com/1/post/2015/05/get-ready-to-record-into-your-daw.htmlJust like in the Recording Phase, taking time to get things right at setup will greatly benefit your workflow and final product.
Here are some considerations when setting up your next DAW session:

  • Naming & Location of Files
  • Sample Rate & Bit Depth
  • File Type
  • Hardware Settings & Buffer Size

It can be tempting to just dive in, hit the record button and get that audio captured so you can get your teeth into post-production.

As we saw with the recording, time spent getting things right at the start is hugely important and will invariably save you time later on.

In this post I will go into further detail about the project checklist as well as giving some tips on setting up a simple template in Pro Tools.

File Naming & Location

As simple and boring as it seems, naming your files correctly is a really important step.

You might think that you will remember 'audio 3' and 'song 2' because its just a quick project but i guarantee that before long you will be digging through files to try and recover that perfect solo buried in similar file names.

The two main file names and locations you want to worry about are the Project Setup and the Tracks.

Whatever logic works for you is fine but for example you might save your projects under type (Jam, Songwriting, Albums, EP's etc) followed by Artist name and finally the Song name.


When naming your tracks, do it in such a way that you will be able to look for that instrument with ease later on (Telecaster, Lead, Shaker etc)

Sample Rate & Bit Depth

A good balance between file size and quality is 24 Bit & 48KHz. 

The arguments regarding these two aspects of digital recording are spattered far and wide across the internet, enough so that i won't go in depth here.

My advice is, rather than spend hours debating or researching different arguments on this topic, get a basic understanding, pick a standard, and put that extra time into making music.

Ok, Oversimplification time;
Computers talk in 0's & 1's, On or Off, Blocks, Bits (Digital)
Sound talks in waves, smooth and curved (Analogue)

If Digital is a light switch that can be either on or off, light or dark, then Analogue is the dimmer switch providing a subtle and continuous change between the points of light and dark.

The Bit Depth relates to how closely to the Analogue curve the digital representation follows.

Its like trying to trace a smooth curve on graph paper while only being able to follow the grid lines. The smaller the Grid (Higher the Bit Depth) the closer you can get to the curved line you are trying to follow
The sample rate is simply how many 'snapshots' of the original signal the computer takes. Long story short the higher the sample rate and bit depth the more accurate the digital representation.
However, remember that the higher you set each of these the larger the file sizes you create and at the end of the day if you are ultimately going to CD it will be bounced down to a Bit Depth of 16 and a Sample Rate of 44.1KHz so setting these at their highest isn't necessarily advantageous.

The File Type you are using is also important and should be a 'Lossless' type such as AIFF or WAV and not a 'Lossy' type like mp3.

Playback Engine & Buffer Size
The first thing you should do after setting your Depth and Rate is making sure you have the correct playback engine selected. This will come up under the name of your interface (in my case Saffire as shown above).

The buffer size is related to how much processing your computer can handle and in theory should be set low for tracking which will reduce latency and high for mixing which will increase the amount of plugins you can use. 
A good starting point is 128 Samples

<![CDATA[How To Record and Acoustic Instrument]]>Thu, 30 Apr 2015 13:43:01 GMThttp://sackrecords.com/1/post/2015/04/how-to-record-and-acoustic-instrument.htmlBig Wins

  • Rockin' Performance & Instrument Sound
  • Get the Best Sound With Just Your Mic's
  • Get Your Phase Sorted

Even (if not especially) in this day and age of recording, the ultimate quality of your sound has heaps to do with how your raw tracks come in on tracking day.

Strong Foundations

There is a misguided tendency in the modern world of digital recording to Fix-it-in-the-Mix rather than to capture the sound well where it comes from.

'A Great Photo Needs a Great Shot'
Performance & Sound
There is no way around it, a great sounding mix comes from a great sounding instrument and a great take.
Put the time in at this end of the project to rehearse, re-string and re-skin to make sure that the sound and energy coming out of your instrument or mouth is top notch.
That includes your instrument being in tune.
Know Your Mic's
You don't need an insured mic locker to get a great sound.
If you can only have one mic, make it a large diaphragm condenser. If you can have two, make the other a dynamic, probably a 57 type.

  • Wider Freq Response
  • Clearer Whole Picture
  • More Spill/Feedback

  • Robust
  • Less Spill/Feedback
  • Handle Higher SPL

Get The Best Sound
If you haven't experimented much with Mic Placement you would be amazed at range of sound that can be captured by simply moving a mic around.

First off, pick a mic and placement as a starting point.
As a general rule Condensers are good for Acoustic Guitars, Vocals and other generally quite sources (Although also often used as Drum Overheads) while dynamics are great for Guitar Amps, Kick & Snare Drums and other high SPL sources or sources where you want to keep spill to a minimum.

Polar Pattern
The Polar Pattern indicated on the Microphone tells you how, and in which direction the Mic picks up sound from. You can use these different patterns to achieve the sound you are looking for. 

A good starting point for placement is placing the mic roughly the same distance away from the source as the length of the resonant component of the source.
For example on Acoustic guitar, measure the length of the guitars body and position the mic approximately that distance away from the instrument.
Generally the closer you get to a source the more Bass Response you will get so keep that in mind.