Get some great sounding music

If you’re a fan of great sound or you want to really test the capabilities of your ears or your gear, here are a couple of resources you might love like I do.

HD Tracks

The first one is HD Tracks (, a site specialising in high quality audio recorded (or remastered) at high sample rates and bit depth. Albums from HD Tracks mostly cost around $18 (AUD) and are mostly available as FLAC files, but in some cases there is the choice of lower quality MP3 files. They have a range of new and old recordings including artists and albums like:

  • Norah Jones
  • The Dark Knight Rises Soundtrack
  • Fleetwood Mac
  • Rolling Stones
  • Aretha Franklin
  • Otis Redding
  • Ella Fitzgerald

They also have a huge range of classical and jazz albums that I’m yet to explore.

The albums on HD Tracks range from 44.1kHz/24-bit recordings through 96kHz/24-bit to 192kHz/24-bit recordings. All of these are significant improvements over the 44.1kHz/16-bit standard of CDs. If you’re unsure of what all those numbers mean, here’s a brief, hopefully simple explanation.

All CDs are 44.1kHz/16-bit audio. This means that the recording system takes a snapshot of the sound 44,100 times per second and that snapshot contains 16 “bits” of information which equates to 65,536 pieces of information (like pixels on a TV – the more pieces, the higher the resolution). The theory behind the 44.1kHz sampling rate of CD audio is that humans can only hear up to 20kHz and so CDs are capturing information more than twice as fast as the human ear can detect.

44.1kHz audio is mostly good enough. You will hear some improvement to the smoothness of the sound at 96kHz and possibly at 192kHz, but at 192kHz it’s debatable and may in fact detract from the music due to the processing power required for 192kHz sound. I choose to use 96kHz audio where possible, but am perfectly satisfied with good 44.1kHz recordings too.

Bit depth is a bit different. Just like a high definition TV looks clearer and sharper than a standard definition TV, the bit depth of audio is the same – more bit depth makes the music clearer and sharper. While there are just over 65,000 pieces of information recorded in each snapshot at 16-bit, 24-bit audio records more than 16.7 million pieces. That’s right, we jump from 65,000 pieces to over 16,000,000 pieces – a massive difference and therefore the sound is much clearer and sharper.

If you’d like to know more about sample rates and bit depth, there’s a great article here.

Grammy Awards – Engineering

Just today, I stumbled upon the interesting fact that every year, there is a Grammy award presented to the album with the best audio engineering. The list of past winners includes some albums worth listening to and a few that aren’t worth as much time (from a musical enjoyment point of view), but they are all beautifully recorded and will make your system shine and bring a smile to your face. There is a complete list of recent winners on the following Wikipedia page:

Grammy Award for Best Engineered Album, Non-Classical

As the name suggests, these are non-classical albums (pop, blues, rock, etc.), but there are also classical albums awarded and the list is also available on this Wiki page.

Having gone through my collection to listen to some of my albums on the non-classical list, I can confirm that they sound awesome and there are some great albums on the list by artists like Sting, John Mayer, Quincy Jones, and Ray Charles.

I hope you find some audio gems amongst the Grammy list or on HD Tracks. Happy listening!




Your Natural Equaliser

A few years ago, I had a problem with wax build-up in my ear (I know, it’s a bit gross, but I have no other way to clearly explain this). Anyway, after putting up with muffled sound in my ear for a few days and noticing it getting worse, I figured it was time to go to the doctor. The doctor made short work of the wax and I soon had squeaky clean ear canals again. So what’s that got to do with equalisers?

I noticed something amazing once my ears were cleaned out – I could hear better than ever before. This was not just a case of it seeming better than I was used to after a couple of muffled days, it was a significant improvement to the details I could hear in sounds. It was like everything had been turned up a few notches – I had bionic hearing!

Needless to say, the changes to my hearing soon passed, but it showed me something important – our brain and our ears adapt significantly over time. Having had the volume and detail levels of my hearing suppressed for a few days, my brain had adjusted its sensitivity to various incoming information, particularly in the upper frequencies that were more affected by the blockage. Once the blockage was removed, these higher frequencies remained more sensitive and gave me the sensation of super-accurate hearing.

When it comes to audio, this is important to recognise. If you’re listening to a new set of speakers or a new set of headphones, it’s important to give your brain time to adjust. Case in point was my recent purchase of some new headphones. Having decided to buy the Ultrasone HFI 680s as a closed alternative to my Audio Technica ATH-AD900s, I spent a lot of time listening to the 680s and I soon learned to love their sound. So much so, that returning to my AD900s left me a little underwhelmed – I suddenly really missed the bass of the 680s and longed for that warmer, fuller sound. And then I realised – I had adapted. My brain had said, “Oh, is this the new ‘norm’? I’ll just adjust to compensate.” Returning to the AD900s which have a more detailed, less bassy sound, my brain said “Where’s the bass gone – there’s a big hole in the sound!” But as I listened to the AD900s for longer, the bass gradually returned and my 680s then started to sound overly bassy when I returned to them.

The interesting and amazing thing is that, once I switched back and forth a couple of times, my brain got quicker at adjusting and I found both sets of headphones more enjoyable almost straight away after switching. It’s like the brain creates its own set of situational EQs that it can switch between when it knows you’re using particular earphones, speakers, or listening in a particular environment.

For music lovers, this is important to recognise. When auditioning new gear, be sure to give yourself plenty of time for your brain’s EQ to adjust. Also, be aware that someone else’s opinion on their favourite speakers or headphones will be coloured by their brain’s EQ settings. What they find warm, but detailed, you might find muddy, especially if you’re coming from a different sounding piece of gear. In time you might adjust to the new gear’s sound, but it may also be beyond the range of adjustment. Our brain still needs to be able to differentiate sounds so it doesn’t try to make everything sound equal, just to balance things out to sound how we believe it should.

Generally, I find it takes up to half an hour of listening to really adapt to a new sound depending on how different it is. Make sure you give yourself that time when auditioning or adapting to anything new or you might just miss out on something wonderful!

A Simple Guide to Equalisers

Equalisers are an interesting topic. Many people will steer clear of them while others love the option to customise every song, album, pair of speakers or pair of headphones to their own personal tastes. To help you make an informed decision about whether to use an equaliser and how to use an equaliser, here are some key facts.

Impact on Sound Quality

Firstly, electronic equalisers like those in iTunes and in your iPod will generally reduce the quality of the sound. Depending on the headphones or speakers you’re using and depending on how loud you listen to the music, you may not notice the difference so don’t necessarily write off the use of an equaliser based on this fact – you need to decide based on the merits for and against. For some people, a slight reduction in quality might be worth the improvement to the overall tone of the sound. There are also some ways to use an equaliser that are less likely to impact on the sound quality – I’ll discuss these later.

The main reason that equalisers decrease sound quality is the power required to boost audio output by even a small amount. A 3dB increase to the sound (or selected frequency) actually takes double the power! This generally means that you’re pushing the limits of the in-built amplifier of your computer or portable player which leads to a thing called clipping. Without going into detail, clipping refers to the top of the sound wave being cut off because the amplifier doesn’t have the power to create it. It’s a little bit like accelerating in your car until the engine hits the redline and the power disappears, resulting in a rapid loss of performance and potential damage to the equipment.

To minimise the impact on sound quality, you can employ a technique that’s often referred to as subtractive equalising. All this means is that you drag the sliders down rather than push the sliders up – I’ll explain.

iTunes EQ

See the image on the right? That’s the standard iTunes equaliser panel and it’s typical of most EQ setups where the sliders are sitting in the middle at “0dB” meaning that there is no change to the standard amplification of each frequency. If you move a slider up by 1 “notch”, you’re increasing that frequency by 3dB and asking the amplifier to provide double the power at that frequency. The result is generally distortion around that frequency. It’s much easier to set an EQ by increasing the frequencies you want more of, but you’ve probably gathered that this will hurt the sound quality.

EQ settings from -12dB

Here’s the alternative: rather than increasing the frequencies you want more of, decrease all the rest! The simplest way to do this is to start by dragging all of your EQ sliders down to the bottom (-12dB). At this time you might need to turn up the master volume of your device to get the sound back to an enjoyable level, but this is fine because the master volume doesn’t decrease quality. Now you’re ready to adjust your EQ. Start using the sliders to increase the frequencies you need up to a maximum of 0dB. Avoid going above 0dB for any frequency because it will instantly decrease sound quality.

All sliders increased (Maximum 0dB)

Once you’ve got the sound the way you want it, try to increase the sliders so that the shape of your EQ stays the same, but so that the highest slider is right on 0dB (see the last EQ image).

Note: one slight issue you might face with this technique is a lack of maximum volume. If your headphones or speakers are hard to drive and you need nearly 100% volume to get the sound you want, this technique may lead to insufficient sound levels. If that’s the case, you might want to look into extra amplification or different headphones or speakers.

The final dilemma you might be facing is which slider to increase to get the sound you want. The next section should help…

What Does Each Frequency Change?

So you’ve opened your EQ settings and you’re ready to perfect the sound signature for your ears only, but where to start? Which slider to slide?

In the end, it’s all experimentation for the fine details, but here are some clues about where to start. (I’m using the iTunes frequency points as a reference to make it easy for comparison.) I’d recommend using the following information by listening to a range of tracks that you’re familiar with. See if you can identify what’s “missing” from the sound based on the descriptions below and then add a little at a time to see if it helps.

32 Hz – This is subwoofer territory – the bottom end of the bass range. As much as we all love it, sometimes it’s better to drop it out or leave it flat if your speakers can’t this depth of bass. Bass is difficult for amplifiers to sustain so you can give your amp some breathing room by dropping this away if you can’t hear it anyway.

64 Hz – This is the part of the bass that we feel as much as hear. This is a great frequency to boost if you want to feel a bit more bass vibration. Increasing this will give your music more kick at the bottom end.

125 Hz – This is musical bass. If you’re listening to melodic bass guitar riffs, this is where the action is. It’s a good frequency to increase to emphasise the musicality and accuracy in bass rather than the rumble or the mass of the bass.

250 Hz – Although there’s no simple definition of bass vs midrange vs treble, I think of 250 Hz as the crossover point between bass and midrange. At 250 Hz we’re starting to increase the bottom end of male vocals and the lower range of instruments like guitars. Beware though – increasing the midrange sliders can very quickly make your music sound muffled or “canned” (like it’s being played through a tin can”. I would rarely increase this frequency, but might consider reducing it slightly to open the sound up a bit and make it more “airy” and spacious.

500 Hz – Like 250 Hz, 500 Hz is the realm of muffled sound. It covers male vocals and the middle of instrumentation. It’s impact on most music is a sense of muffling – like you’ve covered everything in cotton wool.

1 kHz – This is the realm of vocals, instruments like guitars and saxophones and the snare drum. It can bring brightness to the midrange, but can also start making the sound a bit tinny.

2 kHz – Around 1 – 2 kHz is where we start venturing into the realm of treble. We’re still in the world of vocals and instruments here, but we’re getting towards the top-end so it’ll cause vocals to sound a bit more nasal and increase the perception of the texture of voices – things like breathiness and raspiness. It also can make sound very tinny.

4 kHz – This is the frequency that’s most prominent in sounds like “sh” and “ssssss”. It’s also a part of sounds like cymbal hits and the upper end of a snare drum’s sound. Adding emphasis to the 4 kHz range can very quickly make music harsh on the ears and unpleasant to listen to. That said, using it carefully can bring clarity and brightness to vocals and percussion.

8 kHz – This frequency is pure treble. This is what is most prominent if you turn the “treble” dial up. It influences the very upper end of sounds like “sh” and “ssss” and it has a major impact on percussion such as snares and cymbals. It is great to use to add brightness to the sound, but can also get uncomfortable if overused.

16 kHz – Given that 20 kHz is considered the upper end of human hearing, this is obviously the pointy end of the treble band. It mostly affects cymbals and similar sounds, but it also picks up the brightness and detail in the texture of certain instruments. For example, increasing the 16 kHz slider (and/or the 8 kHz slider to a degree) will enhance the sound of the plectrum hitting the strings of a strummed guitar. Like the 8 kHz slider, the 16 kHz slider is a great way to add brightness and tends to be more gentle on our ears. That’s not to say it’s not just as loud, but generally it doesn’t sound as harsh. Incidentally, 16 kHz is also at the end of the hearing range where we lose our hearing first so we might appreciate a slight boost to this area as we get older and lose our hearing.

I hope this has helped you to get more from your music collection, favourite speakers or portable player!

Understanding MP3s (and other compressed music) – Part 3… Finale

Welcome to the final installment of my 3 part series of posts about the pros and cons of compressed audio. If you haven’t read from the beginning, it’d be a good idea. Here’s a link: Understanding MP3s (and other compressed music) – Part 1

By the end of Part 2 you hopefully have an understanding of the process of compression (i.e. removing sounds that we theoretically won’t hear) and also the impact that this removal has on the overall “picture” created by the sound. For this final part of the article, you need to keep this concept of a musical “picture” in mind because this final concept is all about the hidden magic within the picture, not the individual, identifiable details.


You might have heard of harmonics before. If you’ve played certain musical instruments (particularly stringed instruments), you might have even deliberately created pure harmonics. If you haven’t heard of harmonics, don’t worry – here’s a short explanation.

Anytime you play an instrument that uses a string or air to create sound (i.e. just about any instrument other than electronic synthesizers), you are creating harmonics. Harmonics are the sympathetic vibrations that occur along with the note that you’re creating. Have you ever run your finger around the rim of a glass to create a musical note? That’s the same concept. Your finger running on the edge of the glass creates vibrations. If you get the speed of your finger movements correct, the vibrations you create, match the natural vibration frequency of the glass. As a result, the whole glass vibrates together and forms a beautiful clear note. Different glasses will vibrate at different speeds of movement  and will create different notes as a result. This is the concept of harmonics.

If you were to walk up to a piano and strike the key known as “Middle C”, you would hear a note – just one single note, but that note will have a quality very different to the same note on another piano or on a violin. The reason for this is the creation of resonance and harmonics. To explain this, I’m going to talk about the note called “A” which is a few notes above “Middle C”. I’m using the “A” because it makes the maths easier.

If you now strike the “A” you’ll hear a single note once again. This time, the note will sound higher than the previous “C”. What’s actually happening though is that your ear is receiving vibrations in the ear and these vibrations are moving 440 times every second (440 Hz). However, there are also other vibrations going on and the majority of these vibrations are directly related to the 440 Hz we began with. As the “A” string inside the piano vibrates, it creates waves of vibration. The loudest of these move 440 times per second, but it also creates other waves moving 880 times, 1760 times, 3520 times per second, etc.

Every note created by an acoustic instrument naturally creates these harmonics which go up in doubling increments (i.e. like 1, 2, 4, 8, 16, 32, etc.) Old synthesizers sounded particularly fake because they didn’t recreate these harmonics and the output sounded flat and lifeless. Newer synthesizers create harmonics artificially and have come closer to the sound of the real thing, but there’s still a degree of difference created by the subtleties that can be created by acoustic instruments. A slight difference in strike pressure on a piano, plucking/strumming strength on a guitar or force of air through a trumpet can create a significantly different tone as a result of the different range of vibrations it creates. All of these subtleties are the “magic” that make music so special and exciting.

A quick note: this blog is not an anti electronic music. Electronic instruments (i.e. synthesizers, drum machines, etc.) can create amazing music which is impossible with traditional acoustic instruments. The discussion of acoustic versus electronic instruments is designed purely to illustrate the importance of keeping harmonics where they were originally intended/recorded.

Harmonics, Subtleties & Compression

In reading the section above, you might have wondered why you’ve never heard these harmonics. You might even choose to put on your favourite CD and try to listen for them. You can actually hear these harmonics if you listen carefully, but the key thing to recognise here is that we aren’t consciously aware of them in normal circumstances. The harmonics and subtleties happen “behind the scenes” of the music and are rarely noticed by the casual listener or anyone who is not actively listening for harmonics.

If you now think back to my previous discussion of compression and the removal of sounds that we theoretically don’t hear, you might see the connection. The first things be “compressed” (i.e. removed) are the harmonics and subtle, quiet sounds that create the finest details and tonal qualities of the music. To the casual ear, nothing seems to be missing, but play the same song compressed and uncompressed through good speakers and you might notice a difference that you can’t quite put your finger on. Here’s another visual example.

The following picture is a hi-resolution (1900 x 1200) desktop wallpaper image provided with Microsoft Windows 7. I’ve used it because it has a certain magic about it in terms of its depth and detail.

The next version of that image is at a lower resolution of 800 x 500 pixels (a bit like a lower bit-rate of compression).

Notice there’s a certain level of the “magic” missing from the second image? It’s hard to put a finger on exactly what’s missing, but the image isn’t as instantly captivating and engaging to the eye. It almost looks flatter somehow – less bright and alive.

Here’s one last version at 600 x 375 pixels, making it even lower resolution and stealing more of the “magic”.

Are you seeing a difference? Don’t worry if you’re not. Go back now and take a close look at the textures of the character’s face and the stitching on his costume. As the resolution drops, so does the detail. See it? That’s exactly what’s happening to your music.

Compressed Music in Real Life

Although it’s probably clear by now that my preference is always for uncompressed music (known as lossless music because no detail/information is lost), it’s not always practical. Understanding compression allows you to choose what suits your needs best. Here are some factors to consider when choosing your level of compression (or choosing no compression):

  • How much space do you have for your music on your computer, device hard drive, iPod, etc? You’ll need to use compression if your space is limited and you want to store a large number of tracks. Here you need to weigh up quality, quantity and space. You can consider increasing storage space, decreasing the quantity of tracks or increasing the compression (and therefore decreasing the quality of the music).
  • Where and how do you listen to your music? If you listen in noisy environments, at very low volume (i.e. background music only) or use low quality speakers/headphones then you might as well use slightly higher compression to maximise the quantity of tracks. The noisy environment issue can be overcome with in-ear earphones and noise cancelling earphones, but the other situations generally mean you can afford to sacrifice quality for quantity.
  • How much does it matter to you? After all, you’re the one doing the listening so if you’re happy with music at 128 kbps that’s all that matters. There’s no such thing as a right or wrong level of compression – it’s entirely up to you.

The best way to decide is actually quite simple. Take a well-recorded track (or two) that you really like and use your music player (iTunes, Windows Media Player, etc.) to compress it in different ways. Next, listen to the different versions on your favourite headphones and/or speakers and decide what you’re happy with. Way up the differences you noticed between the different levels of compression and think about how much space you have to store music and then make a decision.


Compression is a fantastic tool for portable audio and convenience, but if you have no significant space restrictions, I highly recommend sticking with lossless audio (either Apple Lossless Audio Codec – ALAC, Free Lossless Audio Codec – FLAC or Windows Media Audio 9.2 Lossless). You never know when you might upgrade your speakers or headphones and even if you can’t hear a difference now, you might be amazed at the benefits you get with that next pair of speakers or the next set of headphones! Don’t give up the magic of the music unless you absolutely have too!

Understanding MP3s (and other compressed music) – Part 2

Welcome to Part 2 of my series of posts about the pros and cons of compressed audio. If you haven’t read Part 1, it’d be a good idea. Here’s a link: Understanding MP3s (and other compressed music) – Part 1

Wielding the Eraser

I explained in Part 1 that compression means pulling out sounds that we won’t actually hear, but think about this… The music is like a painting that we “see” with our ears. Compressing music is the equivalent to taking an eraser to the Mona Lisa. It’s like saying, “No-one will notice this brush stroke of stray colour or this tiny bit of shading.” Perhaps that’s true and, to a degree, no-one would notice, but at some point the whole painting’s just going to lose something. It’ll lose a little bit of soul. Sure, you might not pick exactly which parts are missing, but you’ll know something’s not right. Here’s an example:

Notice how the sky in the second image looks unnatural and full of lines? That’s because the process of compressing has removed some of the subtle shades of blue and replaced them with wider bands of other shades. For example, let’s number the different shades 1.1, 1.2, 1.3 and 1.4. During the compression process we would replace shade 1.2 with a second band of 1.1 and replace 1.4 with a second band of 1.3. Now that blue sky would be made of bands of shades 1.1, 1.1, 1.3, 1.3. You can see the evidence of this above in the second image.

So looking at the example photos, it’s clear that they’re both the same photo, but if you had to choose one to print and frame, I’m guessing you’d choose the first one because it’s closer to real life and therefore more pleasing to the eye. The same goes for music.

Think of music as a complex bunch of vibrations making a particular range of patterns. Any little detail you remove from those vibrations will permanently alter the overall “picture”. You’ll still recognise the sound or the song, but it won’t actually sound identical to the original.

Let’s talk about the ear again. Remember my description of how we hear? The ear perceives music like the eyes perceive a painting. You take it all in at once, you don’t pick out a particular colour here and a particular texture there, you just see it as a picture. When we compress sound we permanently alter the “picture” as if we had taken to it with an eraser. To our ears, the result is no different to the photo above on the right. It might not be as dramatic (depending on the level of compression), but it’s essentially the same. You don’t notice a loss of individual sounds, you notice a loss of overall quality and realism.

Here’s one final visual version to show you what I mean. The following charts are spectrograms that show sound as colour. The darker the colour, the louder the sound and the higher up the colour appears, the higher pitch the sound is. A bass guitar shows up down the bottom while a violin shows up further towards the top. There are 2 lines in each chart – these are the left and right stereo channels.

Spectogram - lossless

"This is How a Heart Breaks" - no compression

"This is How a Heart Breaks" - moderate compression

"This is How a Heart Breaks" - mid-high compression (128 kbps)

Notice the density of the yellow and orange colours reduces as you get more compression? The more blue you see, the less of the musical “picture” is still intact. You might also notice that there is more variety and clarity in the colours on the top chart and the colours all get more “blurry” as you move down the charts. That’s the effect of averaging things out. If you look at the first spectrogram and then the second, you might notice that the second one looks like a slightly out-of-focus copy of the first one.

By the time we get to 128 kbps, nearly every high frequency sound is removed. That’s because we lose those hearing at these frequencies first and are less likely to notice the missing sound… or at least that’s the theory. The key thing to notice here is that the musical pictures are different. This is the most visual representation of sound that I can provide and it illustrates exactly how the musical “picture” is gradually erased by compression.

In the Final Installment

Now that you know how we perceive sound and how compression works, you’re all ready to read about why compressed music loses its “magic”. In Part 3, I’ll explain a bit harmonics and their role in creating the soul of the music. I’ll also sum up what this all means when it comes to choosing the level of compression that’s right for you.

As always, I hope you’re enjoying this information and I welcome any feedback or questions you might have.

Ready for Part 3?

Understanding MP3s (and other compressed music) – Part 1

Introduction & Context

As a music lover, I want to experience my music in its purest form. The true purest form is live performance, but we can’t always be at concerts so someone created recorded music. Then someone realised that you can’t take a record player or CD player wherever you go so they created compressed audio. There are many different compression formats including MP3, Microsoft’s WMA, Apple’s AAC, Sony’s ATRAC, and Ogg Vorbis. They all have different names and slightly different methods, but the overall concept is the same.

My aim in this series of posts is to explain what happens when you turn a CD into an MP3 or similar compressed format. In most cases, if you put a CD in your computer, PlayStation, Xbox, etc. and “rip” that music to a disc drive or portable music player, there’s a very good chance the music’s been compressed.

Just like it sounds, compressing music is all about squishing the same length of song into a smaller amount of data. A music track of about 3 minutes 30 seconds takes up between 20-30Mb as pure uncompressed audio. That same track can be compressed at “high” quality to about 7Mb. That’s a massive reduction, but you might be wondering what you’re losing to get the file to shrink by two thirds. Over the next few posts I’ll explain the process and the pros / cons of compression in a simple, real-world way so don’t worry if you’re not technically minded – you won’t need to be.

I should add that I’m not a fan of compressing music, but I recognise the need for it if we want portable music so the overall theme of these posts is to understand what you’re sacrificing when you choose compressed music. Once you know what you’re giving up, you can make an informed decision about what you’re willing to sacrifice in order to carry those extra songs. I hope the information is helpful and interesting.

Key Concepts

The Physics of Hearing: To understand the impact of compression you need to understand how we hear sound. The process begins with a sound source (like a musical instrument) that creates vibrations in the air.  These vibrations travel through the air until they hit our ears. Inside our ears is a thin layer of skin that we know as the ear drum. When the vibrations hit the ear drum, it is pushed around and vibrates in time with the incoming sound. Behind the ear drum are some small bones and our inner ear. The bones get pushed by the ear drum and they vibrate accordingly. As the bones vibrate, they continue to pass the vibrations to our inner ear. You can think of the bones in your ears like the string between two tin can telephones – they just carry a simple vibration.

The inner ear receives the vibrations next and the vibrations “tickle” a bunch of nerves which translate the vibration to a new type of signal for our brain. Don’t worry about the final signal to the brain though, just think about the vibrations until they hit the inner ear. These vibrations are chaotic. They aren’t clear and defined with separate little vibrations for the drums and another set of vibrations for the guitar and another set for the singer, etc. No, the vibrations all pile up and create a big mess of vibration.

A single, perfect note looks like this:

Sine Wave Graph

A graph of a perfect note

This type of vibration is impossible to create with a musical instrument (other than a synthesizer) or voice. Here’s the type of vibration created by instruments and voices:

Music Wave Graph

A graph of musical vibrations

Notice the mostly chaotic nature of the vibrations? There are definitely patterns there, but it’s a big mess of different vibrations. What this graph shows us is how our ear drum would move when receiving this music. The higher or lower each line is, the more our ear drum moves. Lines towards the top push our ear drum in. Lines towards the bottom pull our ear drum out. These movements are all tiny (if the music’s not too loud), but enough to send these crazy vibrations through to our ear nerves. The miracle of hearing is that our brain translates this crazy bunch of vibrations into beautiful melodies and harmonies.

Masking: The second key concept to understand is the concept of masking. Masking is the effect of a louder sound making it difficult to hear a quieter sound played at the exact same time. Think about having dinner in a busy restaurant. You might find it difficult to hear what your friends are saying because of the noise in the restaurant – that’s masking. The combined noise of everyone else’s conversations are masking the voice of your friend across the table.

When some clever bunnies wanted to create a way to store music on computers and iPods (or similar devices) they needed to take some data out of our music. The only data in our music is sound, so they had to find a way to take some sounds out of the music. Sounds tricky, yes? That’s where masking comes into play.

Studies showed that people don’t notice when certain individual sounds are removed from the overall musical landscape. In basic terms, if two sounds occur simultaneously, the quieter one can be removed and we don’t really notice. That’s a slight over-simplification, but it sums up the concept. There are very complex mathematical algorithms and formulas that help determine what sounds will and won’t be missed. I don’t even pretend to fully understand those algorithms so I won’t try to explain it. It also doesn’t really matter how the maths works because the key information to understand is that compression involves removing small pieces of the music that you won’t miss (in theory).

End of Part 1

That’s the end of the first section. Hopefully now you understand how we hear and how masking works. In Part 2 I’ll explain how that knowledge applies to compress sound and how it affects what we hear after the compression is done.