Week 6 Assignment

Hello fellow courserians,

I''m Mircea Oculeanu, from Bucharest, Romania. This is the week 6 assignment for Coursera's Introduction to Music Production.

The subject of synthesis is so vast I couldn't dream of discussing it on a blog post - not even in a looong long series :) of blog posts. But, reviewing some of the subjects discussed this week at Coursera I decided it would be of great interest to see what are the differencies and the common things in synthesizers.  So, let's start with a definition. As wikipedia puts it:
"A synthesizer uses electronic oscillators (or occasionally digital samples) to create an 'artificial' sound. Many have controls for mixing and modulating this to make unique timbres. Most have keyboards to control them, but rare examples such as the ondes martenot and trautonium use other systems. 
Synthesisers can be used to emulate the sound of existing instruments or can be used to make completely new ones. They're very versatile. Modern synthesisers will often have additional sound processing modules such as attack and decay control, distortion or multiple oscillators.

Now, abou our synthesizers. The main difference between Loudon's educational synth and the one I've used is the complexity. But the basic principals are there in both of them

So, we've just heard a preset sound, and I wanted to show you the interface.

The basic structure is the same for both synthesizers: they both have oscillators, only this one - reaktor - has 3 oscillators that produce independantly the sounds! And better than that, you can mix those sounds.

What oscillators do? They're the source of any signal produced by the synthesizer.
Another thing that separates those to synths: the educational one emits just a note at a time - mono - and this one - Reaktor - emits many notes in the same time - polysinth.
Another difference between this two synthesizer is the complexity of the Filter section.

This is one of the least understood sections in a synthesizer, mostly because its not just simply attenuates, it plays a far more important role in a synth. The filter is also called a VCF - meaning Voltage Controlled Filter - and that means that the filter moves over time. Most common examples are wow- wow tipe of sounds, or the vibrato of human voice.
The filter section is placed between the oscillators and the effects; it sculpts the oscillators’ basic sounds - that can be harsh and very bright.
Several modulation sources are available: two ADSR envelope generators, a recordable envelope, and two LFOs combined with a key-scaler that provides four independent control points and four freely assignable MIDI controllers. There's no need to differentiate the two synthesizers here, the eductional one has one ADSR envelope generator, the Reaktor has two but it also have an  additional posibilty, the recordable envelope.
The Reaktor has also an aditional block called Global, where the voice allocation of the synthesizer can be controlled, providing polyphonic and monophonic modes; by selecting the unison mode all available voices are set to the same pitch (as in a monophonic synth), but each one is slightly detuned. This results in waveform interference and a thick, chorus-like sound. Monophonic modes also provide portamento. Parameters determine the master pitch shift and MIDI pitchbend range, and adjust global tremolo and vibrato. Voices’ position within the stereo field can also be adjusted. These settings add enormously to the posibilities u can use with this synthesizer. The pallete of sounds is superb and we've only started with just one preset :).

On top of all these possibilities we have the effects block! You can start from a simple sound or preset and build from there. Ok, I've exagerated with turning those knobs for the demonstration but you've got the ideea :):

These are but a few things that differentiate the educational synth presented by Loudon from this one, more complex from native instruments. What's important is the fact that, in most cases, the basic structure - aka oscillator, filter, lfo's etc - remains the same, the most complex synthesizer only add some blocks to this scheme.
I must admit I'l miss this weekly assignment and I'll miss learning for this course but....hope you've enjoyed it too and thank you for your time and understanding.....

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Week 5 Assignment

Hello fellow Courserians :)

It's been an inspiring and tough ''travel'' but here we are, at week 5 assignment.
I'm Mircea Oculeanu from Bucharest, Romania, and I'll try my best to present you with some of the usages of modulated short dellays  effects (flanger, phaser, chorus).
We've already seen that a simple tiny dellay can be a comb filter in its own only because a small- short - dellay of the same signal creates that notches in the signal. Only that these are static. What if we can move them? It's like we could ''motorise'' them, in a predictable way of course. We can do this differently in the two channels - left and right - so that a mono signal - let's say a guitar - becomes a pleasent and wide stereo sound.
Let's hear one of these modulated dellay effects. I chose flanger because my background as a guitar player draws me to flanger and chorus. But I tried to make things different this time, and listen to what flanger and chorus could do to a drum sound.
First of all, I'll toggle in and out both the effects:

It's really nice and interesting what both flanger and chorus can ad to the drums sound. But let's try to understand how to use them. In ableton live, the flanger is clearly laid out - you can see the sections and clearly enough what they do. Let us remind that the flanger is simply put a comb filter in motion.

On the other hand, chorus uses multiple detuned copies of the sound in order to realize a swirling, wide sound. Let's also remember that chorus modifies the pitch differently on the left and right channels, so the sound will get a more pronounced stereo field.
In the final of this short presentation I'll introduce the chorus on drums in order to hear the spaciousness that it gives to a nice sound drum and how it get out of the way in the mix. I used a demo song in Ableton in order to hear a more complete orchestration and how chorus affects the drums in the mix.

Fact is that all these effects have some kind of learning curve but the most you cand with them is exeperiment and listen. From DAW to DAW they may look different but, in most of them will find sections like LFO, Dry/ Wet etc..
It was a tough week for me and I regret I could'nt purchase the new sound card so that I could've recorded my voice too - with some explanation during the short clips.
Thank you for your time and understanding

Week 4 assignment

The Gate

Hello fellow Courserians,
I’m Mircea Oculenu, from Bucharest, Romania, doing the week 4 assignment for Coursera’s Introduction to Music Production.

I chose a very useful tool that we use in all sorts of situations  - both studio and live environments. And yes, it’s about The Gate – or more appropriate, Noise Gate. What the Noise Gate does is simply letting pass a signal only if it's above a trehsold and below the ''musical signal'' but it should be used with much care. Basically I’ll try to explain the use of gate in the guitar department J but that’s very similar to the use for voices or drums etc..  As a guitar player you are used to having some noise somewhere along the chain. A cable, the hiss from you amp or the poor used effect will ad to the amount of noise you’re hearing. Any guitarist will raise the ‘’Drive’’ value on his/ her amplifier and that’s usually when the hiss will become more apparent. Thing is that with all this noise/ hiss, when the guitar player hits a chord the signal becomes much more powerfull and will ‘’mask’’ the hiss.  That’s where the gate makes it’s entrance.  Because with a gate you can set a threshold somewhere above the level of the noise itself and bellow the signal of the guitar – that the range of gating. Setting this threshold means expiramenting, because setting it too high will also cut into material you want it be heard. Of course u can use it creatively, experiment with what you hear but, for now, I’d like to show you how I’d use it if, for some reason, I have a guitar recording with a bit of noise.

First let’s see what the gate does to another signal – an electric piano -. Modifying the threshold you can hear that, if pushed too far, the gate begans to take out some of the useful, musical sounds too J.

Let’s hear the guitar now, without a gate – it’s bypassed,  and in the second part with slightly different settings of the gate:

As you noticed, when enabled the gate silenced the 2,3 seconds when the guitar didn't produce a chord , 2,3 seconds that you could hear when the gate was not enabled - hiss and a bit of electrical noise. 

You can also use the gate to remove some unwanted parts of your playing - notes touched by mistake etc..

Thank you for your time and understanding!

Week 3 asignment

Hello fellow Courserians,

I’m Mircea Oculeanu from Bucharest, doing week 3’s assignment for Coursera’s Introduction to Music Producation.

In the following few minutes I’ll try to explain the concept and the practicalities of automation in Ableton Live. First of all, why do we need automation in any D.A.W.  Automation is where you decide a parameter – any parameter – should follow a certain pattern which – depending on the D.A.W. you use  - you can draw this pattern. These automated patterns of the parameters could lead to immensly creative sounds but there also used for simple tasks too – let’s say adjusting volume in certain regions of the audio.

I’ll go back to Ableton now, and try to demonstrate how it’s done. You basically  draw a pattern that Ableton reads and follows and we are talking about volume for a track but you cand also do the same with the cutoff in a synthesizer and so on.

I tried to do things as basic as possible because I read in the forums that many of ‘’us’’ courserians J are really beginners in this beautiful domain – music production - . So I’ll try to show you the steps to automate the volume envelope of a basic track.

As you could see, I’ve played a few notes with the keys in my laptop – my midi controller is coming next week J - and. After pressing Tab I’ve changed into session view because the mixer there is more handy. You have one global recording button in Ableton that you must hit and, while playing those few notes again I’ve modified the volume with its respective fader.  Press Tab again to return into arrangement view and you’ll see your automation. A simple and efficient implementation.

That’s a basic presentation for this kind of operation – the automation – but you must note that you can do this with any knob/ function in the modern D.A.W.’s. You practically make a knob/ function move over time in order to obtain variation in sound, volume, envelope, etc, anything you can imagine. Of course, there are several different implementations of the process. Even in Ableton there’s another way of ‘’drawing’’ an automation that the one I’ve just showed you. In fact instead of using the mouse to draw the volume fader you can do the same thing with one knob from your midi controller.  There are certain D.A.W.’s that have separate knobs for recording automation – on every track. Or, you don’t have to record it live – as you’d do on a mixing board but you can simply grab a pencil – the function in the respective D.A.W. – and draw the envelope you want to create.  For any of these scenarios you’re left with the possibility to edit anyone of these envelopes later in the process.

Thank you for your time and understanding.

Week 1 asignment

Hello Courserians,

I'm Mircea Oculeanu from Bucharest, Romania. I'm doing the week 1 asignment, recording an electric guitar or bass without an amplifier example for Coursera's Introduction to Music Production. We'll look into ways to do this when we are supposed to do it without an amplifier.

First of all, let's recall what a microphone does in a similar situation. As an input transducer, the microphone translates the movements of the air into an electric signal. Though the principles are different, the pickup of an electric guitar does quite the same with the oscilations of the guitar strings, namely it produces an electric signal.

image from HowStuffWorks.com

We'll use microphones in all sorts of situatians but the same aplies to recording guitars. Why should we be interested in recording the guitar - be it electric or bass guitar - directly into a line input? Well,  not long ago, the primary means of recording the instrument was to plug it into an amplifier and mike the cabinet; everything else was seen as a poor substitute for the sound of a real amplifier. And there were some reasons for this :) too, back at the time. If you're still not convinced by the sound aquired this way, you can get your guitar recording through a real amp later in the process in order to regain some of the dynamics you feel you're losing by recording directly. And, speaking about home studios it seems obvious why we're interested in recording a guitar direct in our D.A.W. - let me mention just the neighbours.....
Let's now remember that the pickup of an electric guitar is also a transducer, only it's functioning in a different way, that is it converts the oscilations of the guitar strings into voltage variantions. But, based on this principle, the signal is too weak to get it directly into your D.A.W. and, more important, the impedance is too high for the majority of the inputs available into any home studio.
In this situation we'll probably use an audio interface:

I chose this one simply because it's the one I've got and use. As you can see there are 2 inputs on this interface, one with phantom power - witch we don't need right now - and another one for guitar. It also has a ''special'' button tha allows you to monitor directly what you're playing, in real time. Did anyone say ''real time''? Exactly, we are hear facing one big problem of direct recording! The latency. Witch can be simply anoying or distracting and can get you out of the groove. The latency is simply the time necessary for the signal to complete the complete path of our workflow, namely the time necessary for the electric signal of the guitar to get through tha audio interface - that amplifies it to the line level and also converts it from analog to digital - futher on into the computer - and into the D.A.W. - and back till it reaches the speakers or the headphones. With all the eforts made with fast processors and so on the latency still exists, and there are couple of things you can do to diminish it but that's the subject for another presentation: :) so let's stay focused. In case your audio interface does not have the direct monitoring function we can use :

Ta daaaa!

That's the idea that the engineers came up with: the D.I. box. 

What this thingie does is very important because, first of all, makes the transition from the 1/4 TS cable - that can be very short in order to keep the level of noise down - to the balanced XLR cable that gets to the audio interface or to the mixing board. The second and of the same importance: it matches the impedance with the line input of your mixer. So we can now take the XLR out and plug it to our audio interface or directly to the mixing board. That means we can use a longer cable - the XLR cable will not increase noise and the signal loss will be minimum. Some of these D.I. boxes have a through output - parallel output - so you can monitor your playing in real time. As Loudon beautifully puts it in his video this parallel output lets you even plug into a real amp in order to hear the sound, but that's just the case when your neighbours are friendly :)...

And don't forget or I'd rather say never forget going all the way down with the gain knob when you plug or unplug the instrument or any of the cables! it really can heart your ears :)...

I hope u Courserians enjoyed my little example and I 'm waiting forward to hear from you.

Thank you