#impmooc

[firefly2552025]

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Lesson assignment for coursera's IMPOOC course. (Introduction to Music Production)

[musicproductionstudies]

The song used for the video I created for the 3rd assignment for the course Introduction to Music Production.

Take a look at the original post containing the video.

[tapelegaudio]

Week 6: Synths

For this assignment, I'm going to take you though the main areas of the Subtractor synth in Propellerheads Reason. 

Subtractor is a simple subtractive synth, but it has a lot of other features.  If you can play around with Reason, it's worth digging deeper into it.  Also, if you do get your hands on it, make sure to hit the Tab key.  Cool things happen when you hit Tab.

I recommend viewing this at a higher resolution (720p) in full screen mode.  Just to see the detail of what we are talking about here.

Equipment used:

Heil PR40 mic (with pop filter)

dbx 286a mic preamp / processor

Mackie Onyx 1220i firewire mixer (the audio interface)

Software Used:

Propellerheads Reason

Screenflow (the screen capture software)

Audio Hijack Pro (recording both the output of Logic and my input from my audio interface)

[oldtype-blog]

IMPMOOC Lession 5 Assignment - Reverb

Hi, My name is SHIM, Wooyoung (SHIM is my family name, not the first name), and I’m from Seoul, Korea.

I’ll explain reverb - comparing algorithmic reverb with convolution reverb for my 5th assignment in IMPMOOC.

Reverb

A reverberation, or reverb, is created when a sound is produced in an enclosed space causing a large number of echoes to build up and then slowly decay as the sound is absorbed by the walls and air. (from wikipedia:reverb)

In musical terms, a reverb is a kind of delay effect creating a sense of space by adding some delayed sound. We can say a reverb is simulating a real space, and there are mainly two ways of implementing reverb effects: algorithmic and convolution reverb.

Algorithmic reverb

Algorithmic reverb creates a reverb from some mathematical formula, like synthesizing sounds from MIDI.

Various algorithmic reverb effecters

There are various formula for reverb effects, and many implementation. Algorithmic reverb depends on user's experiences and senses, because there's no rule or a kind of regulation how to adjust various input factors.

Convolution reverb

In audio signal processing, convolution reverb is a process used for digitally simulating the reverberation of a physical or virtual space. It is based on the mathematical convolution operation, and uses a pre-recorded audio sample of the impulse response of the space being modeled. (from wikipedia:convolution_reverb)

- from http://d3vvl31cy8gagb.cloudfront.net/135_convolution/recordingSetup.gif

various convolution reverb effecters - from http://d3vvl31cy8gagb.cloudfront.net/135_convolution/convolutionReverbs.jpg

Convolution reverb is as good as the its audio samples (impulse response). It is very realistic, because it's on the sampled data collected from the real spaces - jazz clubs, opera halls, etc. But it lacks of some kind of freedom.

Therefore, it is better two use both algorithmic and convolution reverb to create what you want, according to musical instruments and tracks within your music.

References

http://en.wikipedia.org/wiki/Reverberation

http://en.wikipedia.org/wiki/Convolution_reverb

http://en.wikipedia.org/wiki/Convolution

http://audio.tutsplus.com/tutorials/sound-design/how-to-use-convolution-for-reverb-effects/

Reflection

I'm very sorry for lacking of sound examples or video tutorials. Anyway, Thanks for reading : - )

[oldtype-blog]

IMPMOOC Lession 4 Assignment - Dynamic Range Manipulation

Hi, My name is SHIM, Wooyoung (SHIM is my family name, not the first name), and I'm from Seoul, Korea.

I'll explain dynamic range and the many ways producers manipulate dynamic range for my 4th assignment of IMPMOOC.

Dynamic Range

Dynamic range (DR) has various definition according to where is used: for example, DR in a piece of audio gear is the range levels between the noise floor and the distortion.

Figure: Dynamic Range (of audio gears) http://www.analog.com/static/imported-files/tech_articles/figure5.gif

Similarly, we'll call DR here as 'loudness or volume levels between the quietest and the loudest sound we can make under a certain environment.'

DR is very important for music, because whenever a music becomes quiet or loud, listeners tend to pay focus the change, and DR determines how much a music (performance) varies during its runtime. In other words, musical dynamic can control its listeners emotion, and DR means its extent.

DR Manipulation

We can manipulate musical dynamic range either during a live performance or at its post-production stage. Most important part of manipulating the dynamics of a music is that it should be smooth and natural to help the listener to enjoy music without being interrupted.

Manipulating DR should be done in various scale, to orchestrate the control smoothly.

Figure: Manipulation DR in GarageBand http://coreylatislaw.com/wp-content/uploads/2011/09/Sailing-Away-1024x647.png

Macro scale

A music consists of mani tracks, such as the vocal, chorus, session (instruments including guitars, keyboards, drums, ...). On a macro scale, we use a volume fader or automation to adjust the volume of tracks. For example, we can emphasize main vocal track by making it louder over the chorus track.

Smaller scale

In each single track, manipulating dynamics is very effective. If we don't do 'riding the vocal' - adjusting vocal track volume up when the voice get quiet and the level down when the voice gets loud, the listener may feel uncomfortable and lose focus.

Micro scale

We can manipulate individual events in a single track, mostly drum or percussion track. By controlling the transient of each drum hit, various effects can be done.

Reflection

I've found its very hard to explain dynamic range manipulation without demonstration, but I hope you get any useful idea from my article. Thanks for reading.

[tapelegaudio]

Week 2 Assignment: Editing a Region in Logic Pro

For this lesson, I decided to show you how to edit audio.  This video should give you the tools you need to quickly edit a region of audio in Logic Pro, my DAW of choice. 

The tools I used were:

Logic Pro (DAW)

Screenflow (the screen capture software)

Audio Hijack Pro (recording both the output of Logic and my input from my audio interface)

The video was uploaded in HD, so you can make it bigger and clearer if you like.  

This video went a little long, just under six minutes.  Hopefully, you find it compelling and of a good enough quality to make up for that fact.

Equipment used:

Heil PR40 mic (with pop filter)

dbx 286a mic preamp / processor

Mackie Onyx 1220i firewire mixer (the audio interface)

Thanks for watching.

[george-worroll-blog-blog-blog]

Introduction to Timbre

Introduction to Timbre So, for my Introduction to Music Production course on Coursera, there's an assignment where we have to explain some part of our lessons.

Who am I?  My name is George Worroll, which you could probably have figured out on your own.  I'm living in Buffalo, MO, originally from Milford, CT.

Timbre Now we get to the interesting part.  The above description would be adequate for only the simplest of sounds- a sine wave.  A sine wave is a simple pressure wave that goes up to the peak, and back down again, and back up, in a gentle curve.  This wave can be described with nothing more than a single frequency value and an amplitude value. A sine wave doesn’t do much, though.  While sometimes useful musically, if this were all we had, music would be boring.  The only ways to change the sound would be to change the pitch, or change the amplitude.  We couldn’t make a sound at the same volume and same pitch sound any different.   What can be done, is introducing additional frequencies into the sound wave.  The distance between the peaks, thus the frequency, remains the same.  But the additional frequencies introduce smaller peaks.  Instead of up, down, up, down, up, down... you end up with something more like “up, down a bit, up a bit, down all the way, up a bit, down a bit, up all the way”.  This creates a different pattern of vibration when it reaches your eardrums.  The lowest frequency present, or the fundamental, is perceived as the pitch. Obvious question- Can you add any frequencies you want to and get this effect?  Well, strictly speaking, yes.  You can add a 50 hz wave to a 100 hz wave, and you will hear it as a 100 hz pitch that sounds somewhat different from a 100hz sine wave.  However, most frequencies will not sound very good.  How can you know which frequencies work well together? The Harmonic Series And now, I tie it together.  The Harmonic Series, also frequently referred to as the Overtone Series is a series of frequencies, consists of all frequencies that are an integer(whole number) multiple of the fundamental.  These frequencies tend to complement each other well- Western harmonies are based on essentially the same thing.  Adding frequencies in this series tends to sound musical, so it is used as the basis for pitched instruments like clarinets and saxophones.  It is important to note that many instruments do depend on inharmonic overtones, especially percussion.  But the general rule is that harmonic overtones dominate the sound of most pitched instruments. The relative strength of each of the frequency components of a sound wave determines the timbre.  As an example, the saxophone and clarinet are very similar instruments, yet they sound very different.  The reason for this is that the clarinet is shaped in a way that supresses even numbered overtones.  This creates a sound that is closer to a pure sine wave in comparison to the more complex sound of a saxophone. References: (1) http://oxforddictionaries.com/us/definition/american_english/timbre (2) http://oxforddictionaries.com/us/definition/american_english/amplitude General background: http://en.wikipedia.org/wiki/Timbre http://en.wikipedia.org/wiki/Harmonic_series_(music)

[jamesbusiness-blog]

IMPMOOC Week 1 Lesson: Cables and Connectors in the Studio

Hey everyone, I'm Warren from Louisville, KY and I've been producing my own electronic/hiphop music under the name James Business for just over two years now. I also play bass in a local band and have done some limited recording with them.

I'll be doing my first lesson on cables and connectors. Although, this component of the studio is often overlooked it truly is one of the most crucial elements in the production of any quality recording. It is essential that you invest in high-quality cables to ensure performance and avoid malfunctions and other issues that so often arise when dealing with cheaper cables. There are several standard cable/connector types you will find in most any studio: TS - This is your standard quarter-inch cable or instrument cable. It is a single conductor cable and consists of three parts: the tip, the sleeve and the quarter-inch jack. These cables are used mainly to connect instruments to a preamp or to connect one piece of gear to another in the studio. The main drawback to TS cables is that, because they are unbalanced, noise can easily enter the signal and distort its quality. For this reason when using one you should always try to use as short a cable as you can. The shorter the distance the signal must travel, the less chance noise will affect the signal. TRS - Although, these cables are very similar to the unbalanced TS cables they differ in several important ways. TRS cables consist of a quarter-inch jack, tip, sleeve and a ring around the jack. Unlike TS cables however, they are two conductor cables, which means they send a balanced signal and are far less susceptible to noise and interference. You usually find TRS cables being used as headphone jacks or to send balanced signals across longer distances than a TS cable would allow. XLR - This is the standard mic cable and like the TRS cable it consists of three connectors and sends a balanced signal that rejects noise. XLR connectors are prong-like and lock into the back of the microphone for increased stability. You will use XLR cables in any situation where you want to send a signal across a long distance. Often the issue arises where you need to convert a TS cable with an unbalanced signal to a TRS or XLR cable with a balanced signal. The piece of equipment used for this purpose is called a direct box. These are extremely useful in any studio but especially on stage in a live performance setting. They allow you to input a short-run of an instrument cable and output a longer-run of one of the balanced cables which provides a lot of flexibility for musicians and engineers and ensures the quality of their sound.   Aside from these three main varieties, there are many other types of cables you might find in a studio. Some common ones are eighth inch stereo cables which are mostly used for headphones and RCA cables which most of you will be familiar with as the colored cables you plug in the back of your television. MIDI, Toslink, SPDIF, Firewire, USB, etc. are all cable types you will encounter as a producer as well. The most important thing you can do is educate yourself on the characteristics of each type of cable so you will be able to make the best decisions on how to appropriately use them and get the most out of your gear.