For any musician who's spent a lifetime practicing learning your craft, this might not be something that you'll want to read. Forbes recently compiled a list of
the world's top 10 money making DJ's. While still not as much as the
current pop stars, some DJ's are sure raking in a lot of cash, but
that's thanks to a global EDM industry that is now worth an estimated $4
billion.
Here's the list:
1. Tiesto - $22 million
2. Skrillex - $15 million
3. Swedish House Mafia - $14 million
4. David Guetta - $13.5 million
5. Steve Aoki - $12 million
6. Deadmau5 - $11.5 million
7. DJ Pauly D - $11 million
8. Kadkade - $10 million
9. Afrojack - $9 million
10. Avicii - $7 million
Keep in mind that their income is from touring (Tiesto makes about $250k a night), endorsements, and merch.
Wednesday, August 8, 2012
Introduction to Digital Audio
Digital audio at it’s most fundamental level is a mathematical representation of a continuous sound.
The digital world can get complicated very quickly, so it’s no surprise that a great deal of confusion exists.
The point of this article is to clarify how digital audio works without delving fully into the mathematics, but without skirting any information.
The key to understanding digital audio is to remember that what’s in the computer isn’t sound – it’s math.
What Is Sound?
Sound is the vibration of molecules. Mathematically, sound can accurately be described as a “wave” – meaning it has a peak part (a pushing stage) and a trough part (a pulling stage).
If you have ever seen a graph of a sound wave it’s always represented as a curve of some sort above a 0 axis, followed by a curve below the 0 axis.
What this means is that sound is “periodic.” All sound waves have at least one push and one pull – a positive curve and negative curve. That’s called a cycle. So – fundamental concept – all sound waves contain at least one cycle.
The next important idea is that any periodic function can be mathematically represented by a series of sine waves. In other words, the most complicated sound is really just a large mesh of sinusoidal sound (or pure tones). A voice may be constantly changing in volume and pitch, but at any given moment the sound you are hearing is a part of some collection of pure sine tones.
Lastly, and this part has been debated to a certain extent – people do not hear higher pitches than 22 kHz. So, any tones above 22 kHz are not necessary to record..
So, our main ideas so far are:
—Sound waves are periodic and can therefore be described as a bunch of sine waves,
—Any waves over 22 kHz are not necessary because we can’t hear them.
How To Get From Analog To Digital
Let’s say I’m talking into a microphone. The microphone turns my acoustic voice into a continuous electric current. That electric current travels down a wire into some kind of amplifier then keeps going until it hits an analog to digital converter.
Remember that computers don’t store sound, they store math, so we need something that can turn our analog signal into a series of 1s and 0s. That’s what the converter does. Basically it’s taking very fast snapshots, called samples, and giving each sample a value of amplitude.
This gives us two basic values to plot our points – one is time, and the other is amplitude.
Resolution & Bit Depth
Nothing is continuous inside the digital world – everything is assigned specific mathematical values.
In an analog signal a sound wave will reach it’s peak amplitude – and all values of sound level from 0db to peak db will exist.
In a digital signal, only a designated number of amplitude points exist.
Think of an analog signal as someone going up an escalator – touching all points along the way, while digital is like going up a ladder – you are either on one rung or the next.
Let’s say you have a rung at 50, and a rung at 51. Your analog signal
might have a value of 50.46 – but it has to be on one rung or the other
– so it gets rounded off to rung 50. That means the actual shape of the sound is getting distorted. Since
the analog signal is continuous, that means this is constantly happening
during the conversion process. It’s called quantization error, and it
sounds like weird noise.
But, let’s add more rungs to the ladder. Let’s say you have a rung at 50, one at 50.2, one at 50.4, one at 50.6, and so on. Your signal coming in at 50.46 is now going to get rounded off to 50.4. This is a notable improvement. It doesn’t get rid of the quantization error, but it reduces it’s impact.
Increasing the bit-depth is essentially like increasing the number of rungs on the ladder. By reducing the quantization error, you push your noise floor down.
Who cares? Well, in modern music we use a LOT of compression. It’s
not uncommon to peak limit a sound, compress it, sometimes even a third
hit of compression, and then compress and limit the master buss before
final print.
Remember that one of the major artifacts of compression is bringing the noise floor up! Suddenly, the very quiet quantization error noise is a bit more audible. This becomes particularly noticeable at the quietest sections of the sound recording – (i.e. fades, reverb tails, and pianissimo playing.)
A higher bit depth recording will allow you to hit your converter with more headroom to spare and without compression to stay well above the noise floor.
Sampling rate is probably the area of greatest confusion in digital recording. The sample rate is how fast the computer is taking those “snapshots” of sound.
Most people feel that if you take faster snapshots (actually, they’re more like pulses than snapshots, but whatever), you will be capturing an image of the sound that is closer to “continuous.” And therefore more analog. And therefore more better. But this is in fact incorrect.
Remember, the digital world is capturing math, not sound. This gets a little tricky, but bear with me.
Sound is fundamentally a bunch of sine waves. All you need is at least three point values to determine a sine wave function that crosses all three. Two will still leave some ambiguity – but three – there’s only one curve that will work. As long as your sample rate is catching points fast enough you will grab enough data to recreate the sine waves during playback.
In other words, the sample rate has to be more than twice as fast as the speed of the sine wave in order to catch it. If we don’t hear more than 22 kHz, or sine waves that cycle 22,000 times a second, we only need to capture snapshots more than 44,000 times a second.
Hence the common sample rate: 44.1 kHz.
But wait, you say! What if the function between those three points is not a sine wave. What if the function is some crazy looking shape and it just so happens that your A/D only caught three that made it look like a sine wave?
Well, remember that if it is some crazy function, it’s really just a further combination of sine waves. If those sine waves are within the audible realm they will be caught because the samples are being grabbed fast enough. If they are too fast for the our sample rate it’s OK, because we can’t hear them.
Remember, it’s not sound, it’s math. Once the data is in, the computer will recreate a smooth continuous curve for playback, not a really fast series of samples. It doesn’t matter if you have three points or 300 along the sine curve – it’ll still come out sounding exactly the same.
So what’s up with 88.2, 96, and 192 samples/second rates?
Well, first, it’s still somewhat shaky ground as to whether or not we truly don’t perceive sound waves that are over 22 kHz.
Secondly, our A/D uses a band-limiter at the edge of 1/2 our sampling rate. At 44.1, the A/D cuts off frequencies higher than 22 kHz. If not handled properly, this can cause a distortion called “aliasing” that effects lower frequencies.
In addition, certain software plug-ins, particularly equalizers suffer from inter-modular phase distortion (yikes) in the upper frequencies. The reason being, phase distortion is a natural side effect of equalization – it occurs at the edges of the effected bands. If you are band-limited to 22 kHz and do a high end boost, the high end brickwall stops at 22 kHz.
Instead of the phase distortion occurring gradually over the sloping edge of your band, it occurs all at once in the same place. This is a subject for another article, but ultimately this leaves a more audible “cheapening” of the sound.
Theoretically a 16-bit recording at 44.1 smpl/sec will have the same fidelity as a 24-bit recording at 192. But in practicality, you will have clearer fades, clearer reverb tails, smoother high end, and less aliasing working at higher bit depths and sample rates.
The whole digital thing can be very complicated – and in fact this is only touching the surface. Hopefully this article helped to clarify things. Now go cut some records!
The digital world can get complicated very quickly, so it’s no surprise that a great deal of confusion exists.
The point of this article is to clarify how digital audio works without delving fully into the mathematics, but without skirting any information.
The key to understanding digital audio is to remember that what’s in the computer isn’t sound – it’s math.
What Is Sound?
Sound is the vibration of molecules. Mathematically, sound can accurately be described as a “wave” – meaning it has a peak part (a pushing stage) and a trough part (a pulling stage).
If you have ever seen a graph of a sound wave it’s always represented as a curve of some sort above a 0 axis, followed by a curve below the 0 axis.
What this means is that sound is “periodic.” All sound waves have at least one push and one pull – a positive curve and negative curve. That’s called a cycle. So – fundamental concept – all sound waves contain at least one cycle.
The next important idea is that any periodic function can be mathematically represented by a series of sine waves. In other words, the most complicated sound is really just a large mesh of sinusoidal sound (or pure tones). A voice may be constantly changing in volume and pitch, but at any given moment the sound you are hearing is a part of some collection of pure sine tones.
Lastly, and this part has been debated to a certain extent – people do not hear higher pitches than 22 kHz. So, any tones above 22 kHz are not necessary to record..
So, our main ideas so far are:
—Sound waves are periodic and can therefore be described as a bunch of sine waves,
—Any waves over 22 kHz are not necessary because we can’t hear them.
How To Get From Analog To Digital
Let’s say I’m talking into a microphone. The microphone turns my acoustic voice into a continuous electric current. That electric current travels down a wire into some kind of amplifier then keeps going until it hits an analog to digital converter.
Remember that computers don’t store sound, they store math, so we need something that can turn our analog signal into a series of 1s and 0s. That’s what the converter does. Basically it’s taking very fast snapshots, called samples, and giving each sample a value of amplitude.
This gives us two basic values to plot our points – one is time, and the other is amplitude.
Resolution & Bit Depth
(click to enlarge)
In an analog signal a sound wave will reach it’s peak amplitude – and all values of sound level from 0db to peak db will exist.
In a digital signal, only a designated number of amplitude points exist.
Think of an analog signal as someone going up an escalator – touching all points along the way, while digital is like going up a ladder – you are either on one rung or the next.
Dynamic range versus bit depth (resolution). (click to enlarge)
But, let’s add more rungs to the ladder. Let’s say you have a rung at 50, one at 50.2, one at 50.4, one at 50.6, and so on. Your signal coming in at 50.46 is now going to get rounded off to 50.4. This is a notable improvement. It doesn’t get rid of the quantization error, but it reduces it’s impact.
Increasing the bit-depth is essentially like increasing the number of rungs on the ladder. By reducing the quantization error, you push your noise floor down.
(click to enlarge)
Remember that one of the major artifacts of compression is bringing the noise floor up! Suddenly, the very quiet quantization error noise is a bit more audible. This becomes particularly noticeable at the quietest sections of the sound recording – (i.e. fades, reverb tails, and pianissimo playing.)
A higher bit depth recording will allow you to hit your converter with more headroom to spare and without compression to stay well above the noise floor.
Sampling rate is probably the area of greatest confusion in digital recording. The sample rate is how fast the computer is taking those “snapshots” of sound.
Most people feel that if you take faster snapshots (actually, they’re more like pulses than snapshots, but whatever), you will be capturing an image of the sound that is closer to “continuous.” And therefore more analog. And therefore more better. But this is in fact incorrect.
Remember, the digital world is capturing math, not sound. This gets a little tricky, but bear with me.
Sound is fundamentally a bunch of sine waves. All you need is at least three point values to determine a sine wave function that crosses all three. Two will still leave some ambiguity – but three – there’s only one curve that will work. As long as your sample rate is catching points fast enough you will grab enough data to recreate the sine waves during playback.
In other words, the sample rate has to be more than twice as fast as the speed of the sine wave in order to catch it. If we don’t hear more than 22 kHz, or sine waves that cycle 22,000 times a second, we only need to capture snapshots more than 44,000 times a second.
Hence the common sample rate: 44.1 kHz.
But wait, you say! What if the function between those three points is not a sine wave. What if the function is some crazy looking shape and it just so happens that your A/D only caught three that made it look like a sine wave?
Well, remember that if it is some crazy function, it’s really just a further combination of sine waves. If those sine waves are within the audible realm they will be caught because the samples are being grabbed fast enough. If they are too fast for the our sample rate it’s OK, because we can’t hear them.
Remember, it’s not sound, it’s math. Once the data is in, the computer will recreate a smooth continuous curve for playback, not a really fast series of samples. It doesn’t matter if you have three points or 300 along the sine curve – it’ll still come out sounding exactly the same.
So what’s up with 88.2, 96, and 192 samples/second rates?
Well, first, it’s still somewhat shaky ground as to whether or not we truly don’t perceive sound waves that are over 22 kHz.
Secondly, our A/D uses a band-limiter at the edge of 1/2 our sampling rate. At 44.1, the A/D cuts off frequencies higher than 22 kHz. If not handled properly, this can cause a distortion called “aliasing” that effects lower frequencies.
In addition, certain software plug-ins, particularly equalizers suffer from inter-modular phase distortion (yikes) in the upper frequencies. The reason being, phase distortion is a natural side effect of equalization – it occurs at the edges of the effected bands. If you are band-limited to 22 kHz and do a high end boost, the high end brickwall stops at 22 kHz.
Instead of the phase distortion occurring gradually over the sloping edge of your band, it occurs all at once in the same place. This is a subject for another article, but ultimately this leaves a more audible “cheapening” of the sound.
Theoretically a 16-bit recording at 44.1 smpl/sec will have the same fidelity as a 24-bit recording at 192. But in practicality, you will have clearer fades, clearer reverb tails, smoother high end, and less aliasing working at higher bit depths and sample rates.
The whole digital thing can be very complicated – and in fact this is only touching the surface. Hopefully this article helped to clarify things. Now go cut some records!
Monday, May 28, 2012
Need to be invented
1. A new speaker technology. We've been listening to recorded and reinforced sound with the same technology for about 100 years now. Sure, the loudspeaker has improved and evolved, but it's still the weakest link in the audio chain. What we need is a new loudspeaker technology that improves the listening experience and takes sonic realism to the next level.
2. A new microphone technology. Something is seriously wrong when the best and most cherished microphones that we use today were made 50 years ago. Just like loudspeakers, the technology has improved and evolved over the years, but it's basically the same in that it's still based around moving a diaphragm or ribbon through a magnetic field or changing the electrical charge between two plates (that's a condenser mic, if you didn't know). There has to be a new technology that takes a giant leap to getting us closer to realism than what we have now.
3. Get rid of the wires. Studios have been pretty successful at reducing the amount of wiring in the last 10 years or so, but there's still too much. We need to eliminate them completely. Think how much different your studio would be with wireless speakers, microphones, connections to outboard gear, etc. Much of this is possible today, but the real trick is to make the signal transmission totally lossless with zero interference.
4. The ultimate work surface. Here's the problem. Engineers love to work with faders and knobs. The problem is that faders and knobs take up space, which changes the room acoustics, and which are expensive to implement. When the faders and knobs are reduced to banks of 8, it gets confusing switching between all the banks needed during a large mix. What we need is a work surface that takes this hybrid to the next level, giving the engineer enough faders and knobs to do the job, yet making it totally easy to look at the banks underneath or above. I realize that the bank concept has been implemented on digital consoles for years, but there's no way to actually view what those other banks are unless you call one up. There has to be a better way.
5. The ultimate audio file format. I've done experiments recording the same instrument at 48k, 96k, and 192k and I can tell you unequivocally that the 192kHz recording won hands down. It wasn't even close. Consider this - the ultimate in digital is analog! In other words, the higher the sample rate, the closer to analog it sounds. We need a universal audio format with a super high sample rate that can easily scale to a lower rate as needed. Yes, I realize it's a function of the hardware, but lets plan for the future, people.
6. The ultimate storage device. Speaking of the future, there are a lot of behind-the-scenes audio people that are quietly scared to death that the hard drives and SSD's of today won't be playable tomorrow. Just as Zip and Jazz drives had their brief day in the sun, how would you like to have your hit album backed up onto a drive that nobody can read? That's a more real possibility of that happening than you might know. We need a storage format that is not only robust and protected, but has a lifespan akin to analog tape (tapes from 60 years ago still play today; some sound as good as the day they were recorded). We just can't guarantee the same with the storage devices we use today.
The Quietest Room In The World
If you've never been in an anechoic chamber, it's literally an unreal
experience. Things are quiet; too quiet. So quiet that it's
disconcerting, since even in the quietest place you can think of, you
can still at least hear reflections from your own movement.
I've always assumed that the quietest anechoic room belonged to either JBL (I was told that they have 3 of them) or the Institute for Research and Coordination In Audio and Music (IRCAM) in France, but according to the Guinness World Records, it's actually at Orfield Laboratories in South Minneapolis. Supposedly the Orfield chamber absorbes 99.9% of all sound generated within, which results in a measurement of -9dB SPL. As a comparison, a typical quiet room at night where most people sleep is at 30dB SPL, while a typical conversation is at about 60dB SPL.
The Orfield chamber is so quiet that no one has been able to stay inside for more than 45 minutes due to the fact that you begin to hear your heart beating, you lungs working, and even the blood coursing through your veins. Some people even begin to hallucinate during the experience. In fact, you can't even stand after a half-hour since you no longer hear the audio cues that you're used to when you stand as the reflections bounce off the floor, ceiling and walls of the environment.
While it's easy to figure out what JBL does with their anechoic chamber, what goes on in an independent one like at Orfield? It seems that the chamber is used by companies like Harley Davidson and Maytag to test how loud their products are. NASA also uses it for astronaut training.
Here's a short video that describes the Orfield anechoic chamber.
I've always assumed that the quietest anechoic room belonged to either JBL (I was told that they have 3 of them) or the Institute for Research and Coordination In Audio and Music (IRCAM) in France, but according to the Guinness World Records, it's actually at Orfield Laboratories in South Minneapolis. Supposedly the Orfield chamber absorbes 99.9% of all sound generated within, which results in a measurement of -9dB SPL. As a comparison, a typical quiet room at night where most people sleep is at 30dB SPL, while a typical conversation is at about 60dB SPL.
The Orfield chamber is so quiet that no one has been able to stay inside for more than 45 minutes due to the fact that you begin to hear your heart beating, you lungs working, and even the blood coursing through your veins. Some people even begin to hallucinate during the experience. In fact, you can't even stand after a half-hour since you no longer hear the audio cues that you're used to when you stand as the reflections bounce off the floor, ceiling and walls of the environment.
While it's easy to figure out what JBL does with their anechoic chamber, what goes on in an independent one like at Orfield? It seems that the chamber is used by companies like Harley Davidson and Maytag to test how loud their products are. NASA also uses it for astronaut training.
Here's a short video that describes the Orfield anechoic chamber.
Music is Life
Now comes research that shows that music, as we have suspected all along, has numerous rewards, from improving performance in school to dealing with emotional traumas to helping ward off aging. These come as a result of the brain biologically and neurologically enhancing its performance and protecting it from the some of the ravages of time thanks to the active participation of the player in the act of producing music.
Nina Kraus's research at the Auditory Neuroscience Laboratory at Northwestern University in Evanston, Il. has already shown that musicians suffer less from aging-related memory and hearing losses than non-musicians. They also found that playing an instrument is crucial to retaining both your memory and hearing as you age, and how well you process all sorts of daily information as you grow older.
It turns out that just listening to music isn't enough though. You actively have to participate as a player in order to receive any of the benefits.
That's as good a reason I can think of to learn how to play an instrument and keep on playing it for life. It's not only good for your spiritual health, but your physical side as well.
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