Explain the usage of the 5 most important synthesis modules: Oscillator, Filter, Amplifier, Envelope, and LFO.

The goal of this tutorial is to explain how the 5 more important synthesis modules work, these would be:

Oscillator, Filter, Amplifier, Envelope, and LFO.

Oscilator
An Oscillator generates a sound. Its task is to create a waveform which will produce a different sound depending on the shape of the waveform.

The oscillator does this continuously. The rate at which it generates each cycle of the waveform is what we hear as pitch. Pitch is measured in Hertz (Hz)

The most common oscilators are:

1. Saw Wave - shaped like the teeth on a saw blade, this produces a very common sharp, biting tone.
2. Square Wave - looks like a (near) perfect square, produces a reedy, hollow sound.
3. Pulse Wave - a variation on the above, the pulse wave is half as wide as a square wave, and has the unique ability to have its width modulated (called ‘Pulse Width Modulation').
4. Triangle Wave - unsurprisingly shaped like a triangle, this sounds somewhere in between a saw wave and a sine wave.
5. Sine Wave - a smooth rising and falling shape (like a horizontal ‘S'), this produces a mild, soft tone.
6. Noise - not exactly a waveform, but a source of sound produced by a certain colour of noise.

Filter
A Filter is used to remove frequencies from the waveform so as to alter the timbre.The filters can be labeled VCF (Voltage Controlled Filter) or DCF (Digitally Controlled Filter) in our synthesizer.

After being filtered, a brilliant-sounding sawtooth wave can become a smooth, warm sound without sharp treble. 
 
The most common filters are:

1. Lowpass filter: Low frequencies are passed; high frequencies are attenuated.
2. Highpass filter: High frequencies are passed; low frequencies are attenuated.
3. Bandpass filter: Only frequencies within a frequency band are passed.
4. Band Reject filter: Only frequencies within a frequency band are attenuated.
5. Allpass filter: All frequencies in the spectrum are passed, but the phase of the output is modified.

Filter Parameters

Cutoff Frequency

The cutoff frequency, or cutoff, as the name suggests, determines where the signal is cut off. For instance if a signal contains frequencies that range from 20 to 4000 Hz, and a cutoff frequency is set at 2500 Hz, frequencies above 2500 Hz are filtered. The lowpass filter allows frequencies below the cutoff point of 2500 Hz to pass through, unaffected.

Resonance

The resonance control emphasizes or suppresses signals around the cutoff frequency. Resonance is where the frequencies at the Cut-Off position are boosted (made louder).

• When a little resonance is introduced, the frequencies around the Cut-Off are boosted a little. 
• With a lot of resonance, the frequencies around the Cut-Off are boosted a lot while the lower frequencies will start to diminish a little. 
• When too much resonance is applied, the frequencies around the Cut-Off are so pronounced that the filter will itself start to oscillate and "howl" (like feedback, but in a tuned way).

The overall result of using filter resonance is a change in the basic waveform’s shape and, therefore, its timbre.

Drive

Drive adds an amount of gain to the waveform as it enters the filter an input gain control thus overdriving the filter and distorting the waveform. This waveform distortion changes the timbre of the sound, making it much harsher.


Filter Slope

As you now know , the purpose of a  filter is to cut off the signal at the set cutoff frequency. This cutoff doesn’t happen abruptly but rather at a given slope, which is measured in decibels (dB) of gain reduction per octave.

Put another way, you can define how steep the “cliff” is at the cutoff point by choosing a relatively severe or more gentle slope.





Amplifier


The amplifier module of a synthesizer is responsible for controlling the level—or loudness—of the signal over time. It is the  part of the synthesizer that amplifies the sound that is generated by the oscillator. It normally has a control input which affects the level of amplification and which is normally be driven by an envelope or an LFO  which controls the overall loudness (or amplitude) of the sound over time.

The amplifier section on a synthesizer may be labled as VCA (Voltage Controlled Amplifier) or DCA (Digitally Controlled Amplifier).


Envelope

An envelope generator generates a signal that changes through the length of a sound, normally to control the loudness of that sound. An example of its use is to control the volume of a piano sound. It creates a signal that goes from low to high very quickly (the loud sound happens when the key is pressed), then goes down slowly as the sounds gets quieter.

They are also used to drive other parts of the synthesizer, for example the filter.

The most common envelope has 4 parameters usually   called ADSR Envelope.

 Let us look at what happens to an ADSR envelope when a key is pressed on the keyboard (Before any key of pressed, the envelope is at zero and so there is no sound).

1. Attack - Immediately upon pressing a key, the envelope "opens" from zero to full. This is controlled by Attack time. An Attack time of zero means the envelope goes from zero to full instantly (ie sharp attack). Increasing the Attack time means that this will happen more slowly.
2. Decay - While the key is still pressed, the envelope then drops from full to the Sustain level. This is controlled by Decay time. A Decay time of zero means the envelope goes from full to sustain instantly. Increasing the Decay time means that this will happen more slowly.
    
3. Sustain - While the key is still being pressed, the continuing envelope just remains at the Sustain level. This is set by Sustain level.
   
4. Release - When the key is released, the envelope then drops from the Sustain level to zero. This is controlled by Release time. A Release time of zero means that this will happen instantly. Increasing the Release time means that this will happen more slowly. 

If a key is released during the attack or decay stage, the sustain phase is usually skipped. A sustain level of zero will produce a piano like or percussive envelope, with no continuous steady level, even when a key is held.

Envelope generators are not limited to controlling signal amplitude. They can also control the rise and fall of the filter cutoff frequency or modulate other parameters. In other words, envelope generators can be used as a modulation source or as a “remote control” for a given parameter.


LFO

A Low Frequency Oscillator. It is similar to a normal oscillator except that it outputs very low frequencies (very slow waves) and is used as a control input into another part of the synth. Examples of its use are to create vibrato or tremolo effects.

This oscillator is used only as a modulation source and does not generate any audible signals that form part of your actual synthesizer sound, because it’s too low to be heard. It can, however, affect the main signal by adding vibrato, filter sweeps, and so on.

Without modulation, sounds tend to be boring and fatiguing to the ear. They also sound synthetic, rather than natural, in the absence of some type of sonic modulation. The most obvious type of modulation is vibrato, which is used by orchestral string players to add animation to an instrument’s pitch.



LFO Controls

The LFO generally offers the following controls:

1. Waveform: Allows you to choose the type of waveform (triangle waves and square waves are seen most often). Triangle waves are useful for filter sweeps (slow changes to the filter cutoff frequency) or when simulating an ambulance siren (slow changes to the oscillator frequency). The square waveform is useful for rapid switches between two different pitches (vibratos or octaving, for example).
2. Frequency/Rate: Determines the speed of the waveform cycles produced by the LFO. When set to low values, very slow ramps are produced, making it easy to create sounds such as ocean waves rolling in (when white noise is chosen as the waveform in the main oscillator).
    
3. Sync mode: Allows you to choose between free running (a user-defined LFO rate) or synchronization with an external tempo source (such as a host application).

The LFO can also be controlled with an envelope generator in some synthesizers. As an example of where this might be useful, imagine a sustained string section sound where vibrato is introduced a second or so into the sustained portion of the sound. If this can happen automatically, it allows you to keep both hands on the keyboard.

In certain synthesizers, a simple envelope generator is included for this precise purpose. Often, this envelope consists only of an attack parameter, or occasionally it includes decay or release options. These parameters perform in the same fashion as the amplitude envelope parameters , but they are limited to control of LFO modulations.



LFO Types




In closing

By working on this assignment many doubts I had about the Synthesis modules were clarified, this post is a bit longer that usuall, but i hope this info is usefull for you as it was for me

Thanks a lot for reviwing this tutotial.

Demonstrate two of the three types of modulated short delay effects (flanger, phaser, chorus). Describe how they function and what they are best used for. Be sure to describe the Delay Time, LFO, Feedback and Dry/Wet sections.

The goal of this tutorial is to explain how modulated short delay effects works, in this case, flanger and chorus, and to give you a start point on how to use it in a musical context.

Key Concepts

Comb Filter, is generated by taking the an input signal  and addig a delayed version of it to the original signal.

"The comb filter is a simple recirculating delay line, in which the signal is attenuated (ie. multiplied by some amount less than one) on each cycle. "

"The characteristic of the comb filter is to emphasize the frequency (and its harmonics) of the reciprocal of the length (in time) of the delay line"

Delay Time, it is the ammount of time, usually in milliseconds the original signal will be delayed from its copies. In other words, how long after hearing the original sound the copy of it will be heard

LFO (Low-frequency oscillation), it is an electronic signal which creates a rythmic pulse
and whose frequency is at the bottom of, or below, the audio range. With a LFO we can create pulses that we can apply to our original signal to create a number of effects.

A LFO can generate a variety of useful waveforms, including sine, triangle, square, ramp-up, and ramp-down.The most useful LFOs have an upper frequency limit above 100 Hz and a lower limit below 1 Hz


Feedback, is the ammout of the input signal which is injected back in to the original signal

In the graph we have an input signal (+)  which goes though an effect (A) before reaching the Output, the signal get reinjected with the original input signal.

Dry/Wet,  signal processing units or software have a 'wet/dry' setting which determines the proportion of mix between the original (dry) and 'effected' (wet) signals.

Remember: Dry refers to the unaltered signal.  Wet refers to an altered signal.


How Flanger Works 

Flanger  is an effect that mixes the original signal with a varying, slightly delayed version of the signal. The original and delayed signals are mixed more or less equally. Mixing the original signal with a variable short-term delay creates a series of sharp dips and notches in the frequency response(Comb Filter).

As delay time is shifted up and down by an lfo, a sweeping effect is created.


Speed on the phaser alters the rate at which the comb filter moves through the audio spectrum, depth controls the size of the "teeth" on the comb filtering through the audio spectrum.

Best uses for flanger

Flanging works best on harmonically rich sounds, but it is also strong enough to show up clearly on clean guitar, drums, or even vocals. 


Parameters

Delay Time: Changes the time it takes for the second signal to play after the original. Most times the highest setting will not be more than 20 milliseconds

Depth: This parameter functions as the control for the severity of the changes in pitch.

Width: This parameter is somewhat similar to Depth, but has a noticeable difference to the ear. It controls the speed at which the peaks and valleys of the flange are reached.

Rate: This controls the rate at which the warbles(changes in pitch) repeat themselves. The faster the speed, the faster your audio signal will go through the complete flange process.

Feedback: loops the output signal back into the input, creating a possibly infinite amount of noise.



How Chorus Works

 The idea  is to create clones of the input signal and  then, delay the signal for a few milliseconds, around 15-30ms and also detune (changes the pitch) the copied signal a bit.

Used subtly we get a doubled effect that sounds like many players playing the same part. When pushed to the max, can sound like an intense detuned sound.

Most modern chorus effect unit, be they plugins or hardware based, also spread the effect signal across the stereo field. Sometimes a parameter to control the depth and other behaviours may be encountered.

Best uses for Chorus Chorus is used on many different sounds. Guitar players love it for creating clean sounds. Keyboard and synth players commonly use it to thicken up their sound. For vocals, chorus can help cover up subtle pitch problems. Chorus can also add the illusion of width to a sound, making it appear fuller and wider. Wherever you use chorus, go easy and don't go overboard.


Parameters:

Delay Time:  Controls how long it takes for the second, copied signal to appear. The amount of time is generally kept fairly low—usually between fifteen and thirty milliseconds.

Depth:   Controls the amount of change in modulation or pitch of the sound. The higher the number, the weirder it's going to sound!

Modulation Rate : Controls how fast the pitch will rise and fall. Also sometimes referred to as the LFO frequency. Higher frequencies produce extreme effects. 


In closing

 Try listenging the recordings of your favorite musicians/bands to get a better sense of how to aplly Chorus and Flanger ir a musical context.

For flanger you can listen to Jimi Hendrix´s compositions, you can try with  Voodoo Child

For chorus  you can hear it heavily used in Nirvana´s song Come as you are from , also in the spanish hard rock band Héroes del Silecio, Juan Valdivia(guitar) plays a lot of arpegios and does use a lot of chorus (and also flanger) to thiken up his tone (try Maldito Duende and Sirena Varada) .

Thanks a lot for reviwing this tutotial, hope this was useful for you.

Demonstrate an effective use of a compressor in a musical context (hopefully a piece of your own music).

INTRODUCTION

The goal of this tutorial is to demonstrate how to  use of a compressor in a musical context.

Compression is used to bring down the highest peaks, above the threshold level, leaving the lower levels just as they were. (Downward Compression)

Compression also amplifies quiet sounds by narrowing or "compressing" an audio signal's dynamic range. (Expander)

Most compressor have the following common parameters:

1. Threshold sets the level where compression starts to take effect.
2. Ratio, which sets the degree of compression above the threshold level.  A ratio of 2:1 represents mild compression and means that when the incoming level (above the threshold) rises by 10dB, the outgoing level will only rise by 5dB.  Ratios of 10:1 and known as limiters.
3. Attack, measured in milliseconds and tells the time taken for the compressor to start working once the signal has passed the threshold. A fair amount of attack will give us a more realistic sound whereas removing too much attack can produce a fake sound.
4. Release sets the length of time it takes for the compressor to go back to its normal state once the signal has gone back below the threshold.

DAW: Ableton Live 9, default compressor (Under Audio Effects)

The Video, will hopefully, guide you in the process of adding compression to your creations, do pay attention to the sounds before and after adding compression, sometimes it is useful to use external speakers instead of your laptop's default speakers in order to get a better listening of what's going on.

In closing, Compression is widely used in contemporary music, one of the bands I like that use a lot of compression in their sound is Rammstein, anyhow, try to listen the use of compression in your favorite tunes, it is actually a good exercise.

The tune I used, is of my own creation, I will upload it to soudcloud when I am done working on it.

Hope you have enjoyed the video, thanks a lot for reviewing  this material, good bye for now and good luck with this week's lessons.

Note: If the video looks blurry, please  in the  youtube viewer click on the cog icon and set your resolution to 720 HD (or higher if you prefer )

The tutorial is 5 minutes long, I added a couple of seconds of the final result so that you can listen to it.

Submix practical: Demonstrate the configuration of a Submix in a DAW or physical mixing board.

INTRODUCTION
The goal of this tutorial is to demonstrate how to  create a submix in a DAW.

The concept of a submix is to have a group of tracks routed to an aux track and being able to control the group of tracks with only the selected aux track. Of course we can also add effects to the aux track, and the effects will be applied to all the tracks linked to that aux track.

  
DAW: Apple Logic Pro 9, (I was planning to do the tutorial in Ableton Live 9, but is is quite simple to use groups in Live 9 and route it to an aux track, so I took the decision  to use a more complex DAW)

The video: Will guide though the process of naming the buses, assigning each track to the chosen bus, special attention should be taken in the process of assigning the three Electric Guitar tracks to one single  bus to create a submix

In closing. Submixing is a powerful technique that can help us to control many tracks at the time with the use of one single fader and also we can add effects to many tracks by using a submix.

I would have loved to have used drums for this tutorial but I don't have a hardware mixer yet. Hope you have enjoyed the video, thanks a lot for reviewing  this material, good bye for now and good luck with this week's lessons.

Note: If the video looks blurry, please  in the  youtube viewer click on the cog icon and set your resolution to 720 HD (or higher if you prefer )

Add a software instrument and record MIDI and quantize in your DAW

Introduction


The goal of this tutorial is to show you how to add a software instrument and record MIDI and quantize in your DAW. Including preparing the track(s), adding the instrument, setting the click and countoff, and recording efficiently.

For this assignment I have chosen to create a video, the following is the check list of software I am using:

DAW: Ableton Live 9  https://www.ableton.com/en/trial/
Midi Controller: Virtual Midi KeyBoard http://sourceforge.net/projects/vmpk/
Software Instrument: Syth1   http://synth1.com   http://www.geocities.jp/daichi1969/softsynth/

The video will guide you through the process of setting up the recording preferences, setting up a countoff, adding a software instrument, recording a clip using your computer keyboard as a virtual midi keyboard, adjusting bar length, setting up quantizing and  quantizing a clip, and as a bonus: adding a simple drum clip. 

In closing, Synth1 is a highly CPU/Memory efficient VST Software Instrument which virtually offers unlimited sound options for free, it has an incredible amount of patches and an unofficial manual that will easily guide through the process of creating your own sounds. It is available for Mac and PC.

I didn't have my MIDI Controller available at time of creating the video so my option was to use a Virtual Controller such as Virtual Midi KeyBoard instead, and it actually did a decent job.

Hope this small tutorial was useful. Thanks for taking time to review my post, good bye for now and good luck with this week's lessons.

Edit. I had to upload the video again to youtube, first one was low res, sorry about that. If the video looks blurry, please  in the  youtube viewer click on the cog icon and set your resolution to 720 HD

How To: Recording an Electric Guitar or Bass without an Amplifier

Introduction

So, the main goal of this tutorial is to give a short and clear answer to the question "How do I record my Electric Guitar without an Amp?"

There is a technique known as direct recording which, is few words, is connecting your guitar to the DAW using an audio interface.

So the main issue for me is that I don't yet have a USB mixer or an audio interface that I can use to record the sound of my guitar, but wait a minute I do have a Zoom G 2.1U which happens to have a usb port that I can hook up to my computer.

Ok, so let go ahead an describe the process to record my guitar sound.


The connections schema:

My setup goes like this: From the guitar It run a 1/4 inch TS cable to the Zoom G 2.1U which, as stated before, has a USB connector, from there I run a USB 2.0 cable to my Mac, form the audio out of the Mac i run a stereo cable  which goes to the 5.1 Speakers set.

One thing you have to notice is that I am not using a power source for the Zoom G 2.1U, that is because the USB power will be enough to get the Zoom pedal running, just make sure the power switch is set to off on the Zoom G 2.1U.

So far so good, but I can't still hear anything coming from my guitar, next step is use a DAW so that we can actually hear and record the sound coming from the guitar.

Listening  and Recording our sound.
When recording or playing guitar sound using direct connection I use GarageBand, which is simple and flexible enough for what I intend to do.

First we mus tell GarageBand which is going to be our sound source, so:

Open preferences window: GarageBand  ->Preferences 
Audio/MIDI: Select desired input from drop-down menu, which in our case will be the Zoom G 2.1U
So, that's it for sound input.

For sound output, I do the following:
Open preferences window: GarageBand  ->Preferences 
Audio/MIDI: Select desired audio output from drop-down menu, in our case that'll be System Setting 

Once it is done we need to add a track in GarageBand. This track is going to be were we record the sound coming from the guitar.

Once we add a track we must choose the type of instrument the track is going to hold. In this case we can choose Electric Guitar, GarageBand will add some nice effects and an amp simulator to our track so that we have a start point.

Also we must notice that at left we have a drop down which prompt us "My instrument is connected with" we must make sure is says USB or we will not be able to get any sound coming from the guitar.

On the right side we have another drop down which prompts us what is going to be the output for the sound, System Setting will do the trick.

We can also choose on the bottom left if we want the sound we hear and record in mono or in stereo.

Finally and very important is the I want to hear my instrument as I play and record check box, which must be selected so that we can hear the sound of our instrument in the speakers system attached to the Mac

Then we click on Create and Voilà, we are ready to record our guitar sound without using an amplifier.

In closing, I must let you know that I use this method to record my guitar sound with a backtrack (usually Bass Guitar, recorded in the same way and a midi track with the drums), as you  can see in the image above, which so far has worked just fine for me. 

Also in the guitar track you can tweak your sound using the plugins provided in GarageBand or you even use an Amp  modeler software such as Amplitube, in that way you can try some of the most incredible Guitar Amps and stomp boxes.

Hope this small tutorial was useful. Good bye for now and good luck with this week's lessons.