In-Browser Synthesizer!

Posted on May 15, 2013 by CRAS

Here is a cool page that Murat Ayfer developed that allows you to use your computer keyboard to play sounds out of a web browser. You can also go through and modify various aspects of the waveform, including timbre, time, frequency, duration, envelope and pitch bends. This is a neat and fairly intuitive way to get into understanding how synthesizers can be used, and how you can make your own sounds. There is also a visual display as to how the virtual “strings” would resonate as the sound is being generated. You can also look up presets that other users have made to help you get more of an idea of what is going on.

*Note – currently this page only works through Google Chrome or Safari web browsers. He is working on adding support for other browsers, such as Firefox.

Timbre, (the texture of sound; the thing that makes you distinguish between a guitar and a saxaphone) is our perception of the particular overtonic content of a sound.

This means that a single note you hear has several frequencies playing at once. The loudest frequency is referred to as the note you hear. For example, 110Hz is an A. The other, quieter multiples of the root are its overtones. 220Hz, 330Hz, etc. would be overtones of A.

Tonehack lets you pick these frequencies yourself, and using your cursor, draw how these frequencies change over time.

tonehack.com

Intro to the Basics of Digital Recording

Posted on May 8, 2013 by CRAS

To start off with, audio is our perception of differences in air pressure. Common vibrations, or frequencies, are what our ears use to determine what a sound is, how high the pitch is, and where it is coming from. When recording audio to an analog style medium, sound waves are converted or “transduced” in the microphone capsule. The diaphragm of the mic is pushed, and that changes the active voltage going down the microphone cable. The electronic signal can then be processed as negative or positive movement, instead of physical pressure. Speakers work much in the same way that a microphone does, just in reverse. They take the electronic signal and use that to push the speaker diaphragm in and out, thus creating sound pressure waves that we can hear.

When we are going to record to a digital medium, the electronic signals need to be converted into digital, or binary, code. This is the job of the A to D converter – Analog to Digital. Sometimes these are referred to as ADCs, and then on the back end there are the DACs, or Digital to Analog Converters. In many cases, we call these devices “interfaces”. They are designed to accept an incoming analog signal, and they can then convert it into digital audio that can be worked with in a DAW, or digital audio workstation. Often these interfaces can connect using USB, Firewire, or their own proprietary interface card that must be installed into the computer.

Digital audio is based on taking samples, or “snapshots”, of the waveform. Instead of having a continuous flow of information, like analog audio does, digital audio is just a series of pictures of the audio. Similar to how movies are a series of pictures that are played one after another, digital audio is a series of samples played one after another to give us the impression of true audio.

There are two components to digital audio – sample rate and bit depth. Sample rate refers to how many snapshots are taking of a waveform every second. This directly relates to frequency. The higher the sample rate, the more accurate of a representation the digital audio can present, and the higher the frequency can be recorded. For example, CDs have a sample rate of 44.1kHz, or 44,100 snapshots per second. Using what is know as the Nyquist theorem, the highest possible frequency that can be recorded is 22,050Hz, which is just above the standard range of human hearing.

Bit depth correlates to volume, or amplitude. One bit represents two possible values – either 0 or 1, on or off. If we were recording using one bit quality, we would only hear sound very loud, or completely silent. If we increase the bit depth, we can thus get better resolution in the sound. With two bit quality, there are four possible values – off, quiet, louder, loudest. This way we can start hearing the “dynamics” of audio. CDs use 16 bit depth recording, which allows for over 65,000 levels of volume.

Read more about the Basics of Digital recording here.

Crash Course on Sonic Booms

Posted on May 2, 2013 by CRAS

In the air of our atmosphere, sound generally travels near 1130 feet per second, which is around 770 miles per hour. Various factors come into play when measure the speed of sound, including temperature, humidity, and the material the sound is propagating through.

When something, such as a jet, exceeds the speed of sound, it creates a sonic boom. A sonic boom can be compared to waves crashing on a beach. Watching how water reacts to crafts moving through it, we can see the water be displaced in the shape of a wake, which spreads out and away from the boat. Sonic blasts are very much like that, where the plane is dragging a “wake” of air pressure behind it. As the jet flies by above the speed of sound, we see the plane before we hear it. Then the wave of air pressure behind it crashes into us, and we hear the boom.

Check out this interesting video on sonic booms!

What’s the best piece of advice you’ve ever been given?

Posted on April 30, 2013 by CRAS

Culled from numerous responses, here is a small collection of some of the best tips to improve your audio engineering skills:

It’s all about mic placement. If you want the best sound, start off with a good quality source. Spending a few extra minutes getting every set up correctly will save you time in the editing and mix down process later.
Never say “we’ll fix it in the mix”.
Work smart, not hard. Use technology to your benefit. By creating a session template that is configured with empty tracks, commonly used plug ins and routing already in place, you can save yourself set up time and you won’t lose your creative flow setting up the necessary elements.
Critically listen to your work. Recording and mixing are audio arts, so try giving them a pure listen once in a while. Just like blind people have other heightened senses, shut off some of yours to focus on what’s important – close your eyes and really listen.
A good mix will sound good whether it’s quiet or loud. Turning down the volume can help prevent ear fatigue, and it also makes certain elements stand out. Maybe at lower volumes the high hat will be more apparent, but you otherwise wouldn’t have noticed since the louder sounds mask it at higher volumes.
Spectrum analyzers – Putting one on your master output can help you see what frequencies aren’t represented, and you can see if your mix is bass heavy without having to depend on your room/speaker set up to tell you.
EQ is great but don’t overuse it. Boosting a lot of frequencies on various tracks can lead to congestion and conflicts. Instead of boosting some frequencies, think about cutting something else to avoid clutter. This can free up the aural space and give more room to the elements you want to stand out.
Take a break! Your ears get tired, just like everything else. Make sure to chill out every hour or two during a project. Your ears will be refreshed, you’ll get a new perspective on your mix when you come back, and you might even be able to think of other techniques to implement since you’re not distracted with the task at hand.
A good performance makes the track – not a technically skilled rehearsed rendition. That’s not to say that having skills is a bad thing, but when an artist plays the same performance over and over, sometimes it gets to a point where they are just going through the motions instead of giving a good feel. This ties into an earlier tip – by having templates and everything preconfigured you can get straight to recording without losing the vibe.
Low frequencies fill the audio spectrum and have more power than higher frequencies. Use high-pass filters on tracks that don’t have a lot of low frequency content to free up room. Even some bass elements – kick, guitar, low tom – can be improved through the use of a high pass filter by cutting out the muddy, super low frequencies.

What Did Alexander Graham Bell Sound Like?

Posted on April 29, 2013 by CRAS

Alexander Graham Bell is considered one of the most prolific inventors of modern times. Arguably, the technology that we all use today has been derived from the work the Mr. Bell accomplished in his time. Born in 1847, Bell grew up in a world where some of his most cherished people were deaf – his mother and wife. This clearly drove his studies in a certain direction, and his most notable creation was the patent for the telephone in 1876. He has also been credited with assisting in the founding principles of optical telecommunications, aeronautics and hydrofoils.

While he is widely recognized, most people have probably never heard his voice. Passing away in 1922, his legacy continued in the devices he helped create. Recently scientists working the Smithsonian Institute have been able to uncover his voice through some of the recordings that he made during his experimentations. In late 2011, Patrick Feaster, an Indiana University sound and media historian, worked in conjunction with National Museum of American History curator Carlene Stevens to extract audio from some of Bell’s recordings. Bell’s experiments included recording on various materials, such as metal, paper, glass, wax, plaster and cardboard.

The National Museum of History has a number of Bell laboratory discs, ranging from 4 inches to 15 inches in diameter, showing the various techniques the group used to find the best quality recording medium. By comparison, most modern records released nowadays are 12″ vinyl records. The latest recording they have been able to transcribe was a wax inscription upon a cardboard base, containing a recording of Bell speaking.

You can listen to the recording here or through the Smithsonian Magazine website.

Bio-violin

Posted on April 16, 2013 by CRAS

In 2009, scientist Francis Schwarze and Swiss violin maker Michael Rhonheimer decided that they would try to make a violin out of fungi-treated wood and see how it sounded. While doing research, checking on the health of local trees by using sonograms, Schwarze noticed that some trees hand fungal infections that did not produce rot. While the fungus decreased the density of the wood it was in contact with, most other physical properties of the wood stayed the same.

The team ended up creating four violins – two identical pairs, except that one pair would be treated with a type of fungi and then allowed to “cure” for several months. A fifth violin ended up cracking after it had been curing for a year, and was not used for the experiment.

The wood ended up having much less density, however sonically it seemed to improve vastly. According to Schwarze, treating the wood with fungi artificially recreates the structure of the wood that was naturally occurring during Antonio Giacomo Stradivarius’s lifetime. The Little Ice Age, a period of abnormally cool weather between 1645 to 1715, made trees create more uniform wood. Treating wood with the fungus artificially recreates wood similar to that which was created in the Little Ice Age.

A hearing test was put on at a German convention analyzing forest husbandry. British violinist Matthew Trussler brought in his own $2 million Stradivarius, and played a blind test using that as well as the four specially built violins. A jury of experts, as well as the conference attendees, judged the tone quality of the violins, and the ultimate winner was “Opus 58″ – one of the fungus-infected violins.

You can read more about this article here and here.

Mazda 787B Game Audio Recording

Posted on April 5, 2013 by CRAS

1991 saw Mazda’s win at the 24 Hours of LeMans using their 787B group C racing car. This is one of the DLC cars available in the XBox 360 game Forza 4, which is arguably one of the best racing video games in existence currently. One thing that makes the Forza series stand apart from other racing games is the developer’s attention to detail. Every track was recreated by using GPS measurements and going on and actually walking the track. Every car has upgrade options, and stay true to the real-life versions of their respected cars.

What is even more incredible is the audio. This is a game that I can play without any music, and just listen to the engines roar. The Turn 10 team responsible for typically records engine sounds at Dyno Authority in Seattle, but for this particular car they needed to go on a field trip. With 700 horsepower and no mufflers, this could be one of the loudest cars produced.

Special care was taken into consideration when recording the audio for this car. Even while using high SPL handling mics, they had to add sometimes multiple PADs, or pre-attenuation devices, to ensure a good recording quality without distorting or overloading the signal chain. Watch this behind the scene footage and see how they did it all!

Do I need to break in my new speakers?

Posted on April 3, 2013 by CRAS

There is a lot of talk about whether or not newly bought speakers should be broken in. On one side, speakers are a type of mechanical device and a lot of people compare new speakers to a new car. Most car manufacturers recommend that you drive the car specific ways to break in the motor and mechanical parts correctly. For example, it is recommended to drive at varying RPMs for the first 500 miles with some turbo or high performance vehicles, as opposed to setting cruise control and running it on the highway at the same speed.

The idea behind that is the parts will slowly finishing the machining process and make the final adjustments and seals while they are running in conjunction with all the other parts.

Some people have told me that you want to break in the speakers by playing music through them at low volumes for extended periods of time before really using the speakers at normal levels. Perhaps it has to do with gently breaking in the voice coil and allowing the diaphragm and speaker cone to flex and get into normal playback shape.

Some people have some interesting, if not voodoo-like methods of breaking in their speakers. One internet user suggests

Use pink noise or FM noise, then place the speakers face to face and close together, about a foot apart. Then switch the wires on one of the speakers, black on red, red on black, thus reversing the phase. This is an effective way to break in speakers.

One of our school techs said when he used to repair 15″ JBL speakers, they would take them up on the roof of the shop and run a 60Hz tone through them at pretty decent levels for a few hours.

Other people seem to feel that speaker break-in is unnecessary. With better machining tolerances at the factory, and different materials – like kevlar speaker cones instead of paper – the real need to “break in” a speaker may be more of a psychological process than anything.

Often people may listen to a set of speakers in a store and think they sound great, but when they put the speakers in a new room and atmosphere, their impression of the sound may be different. So perhaps sales people offer this advice of breaking in their speakers to prevent them from returning them immediately upon hearing them for that reason.

The Sound of Stonehenge

Posted on April 2, 2013 by CRAS

A team of scientists have recently reconstructed the soundscape of the iconic Stonehenge circle. Using archaeoacoustics and state-of-the-art sound gear they were able to create impulse responses to define the acoustical behavior in the physical space Stonehenge is comprised of. Since electrical equipment is not allowed in the actual Stonehenge area, they did what they could using balloons and battery-powered field recorders, and matched their results with a WWI era memorial replica of Stonehenge at the Maryhill Museum.

Their results show that the design of Stonehenge lends itself to being a highly reflective sound space, allowing 1 second of reverberation time before full decay. This result is similar to what you would find in a large lecture hall, and is optimal for speech intelligibility. This would have allowed speeches to be very well heard within the stones, regardless of where the speaker was standing. The spacing and angles of the stones helps prevent strong reflections which could cause interference or unwanted feedback at higher volumes.

“The existence of a 1 second Reverberation Time would certainly be noticeable to any person entering the circle. Interestingly, 1s is a typical, optimal RT for a large lecture hall, ensuring good speech intelligibility. Anecdotally, the space exhibited this feature, i.e. speech was clearly audible regardless of speaker and listener position, undoubtedly due to the large number of reflective surfaces surrounding them but also due to the high degree of scattering provided by the interspacing between the stones, preventing any strong reflections from becoming a nuisance when interacting with the direct sound from the speaker.”

This is a very interesting experiment, and could definitely be used to uncover what Stonehenge was originally built for.

Why say “check 1, 2″?

Posted on March 23, 2013 by CRAS

Just about everyone who has seen a live concert is familiar with hearing the phrase “check 1, 2″ or “test 1, 2″ while people are testing the mics. But why is this used instead of some other sort of phrase?

People tend to have varying opinions on this, but one thing is sure, no one likes to hear “check check 1 2″ all night, so is it really useful? There are actually a few reasons behind using this phrase over others, although there are plenty of other phrases that can accomplish a similar task.

First off, let’s look into a few terms related to vocals, and specifically vocal miking and recording.

Sibilance – this refers to the sound produced when using words or phrases with “s” in them, similar to hearing a snake’s hissing sound. This occurs in a range of frequencies that can commonly come off as harsh sounding and scratchy. The “ch” sound in check as well as the “st” in test are both good at revealing potential sibilance, and allowing for the sound guy to compensate.

Plosives – this refers to harder “popping” type sounds. Typically created by a stopping consonant, or a hard consonant sound such as “ck” in “check” and “tw” in “two”. Plosives have the potential of creating thick transients that can set off a feedback loop if the sound system isn’t set just right.

In many cases, having a pop filter or wind screen can definitely fix these problems. However, those are not commonly used in the live sound seen, although they are a mainstay in the studio.

After looking around, I found a few other logical reasons behind this little phrase:

It’s easy to remember, so even someone with stage fright won’t forget (hopefully)
This combination of words represents a good cross-section of the vocal range, with lower frequencies represented by “one” and higher by “check”
Riggers lift on 3, so sound guys mess with the crews by constantly saying 1, 2, but never get to 3.

That’s what I could come up with. What do you think?

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