Synth Basics

Week 6: Synth Basics -- Compare the graphical interface of four different synthesizers. Clearly show where the Oscillator, Filter, Amplifier, Envelope, and LFO sections are.

One can find many VST synthesizers on the Internet to download. Many of these have a unique graphic interface where all the effects can be set, but it is a bit time-consuming for one to find out what a single synth is capable of. Modular design is a common feature: on almost every interface you can find an Oscillator (OSC or VCO), a Filter (FLT or VCF), an Amplifier (AMP or VCA) with an Amplitude Envelope (ENV), a Low Frequency Oscillator (LFO), and often additional Effects (FX). ('VC' stands for 'Voltage Controlled' in the beforementioned abbreviations.)

On the following figures you can find the graphic interface of four synthesizers with the different modules marked with different colours. In general these interfaces look like real hardware as they have a bunch of knobs and buttons and faders to fiddle with, but unlike most real hardware they often have little screens to show how the waveform looks like, just like an oscilloscope. Some of them even have detachable wires to connect different modules, so one you can redefine the order of these modules and so alter the behaviour of the synth.

Fig.1: Artphase 1.5

Fig.2: Linplug's Free Alpha 3

Fig.3: OdO's Purple 2

Fig.4: Additive Table Synth

Source: http://lesitedeburnie.free.fr

On this site there is a list (under the practical name 'The List') where you can find many free synthesizers to download. You should give them a try if you are interested.

Equalization Basics

Week 5: Equalization Basics -- Demonstrate the configuring of an EQ plugin to function like a large format mixing console EQ section. Include instructions showing how to save the setting as a preset in your DAW.

Whenever you have an audio recording of decent quality, you'll still want to adjust its timbre to fit in your mix. One way to do this is using an Equalizer (EQ) plugin in your DAW.

Contemporary mixing boards (even the smallest ones) tend to have a separate EQ section for each channel, which you can customize its sound with. The bare minimum is a Low Level and a High Level controller, but another one for Middle Level is also common. Larger mixing consoles have separate knobs for Low Mids and High Mids. In most of this cases there are additional controllers to adjust the center frequency of the beformentioned two knobs. Often there is a two-state button in the beginning of the EQ section which you can cut the sub-bass frequencies with.

When you start equalizing your recordings in a DAW, you'll want to set it up in a way you can easily adjust the main frequency bands. For this you should make a preset based on an EQ section of a real mixing board, because they are made for being wieldy and efficient. Remember to use the smallest number of EQ bands you can to save computing capacity.

Fig.1: EQ preset based on an EQ section of a mixing board

The first thing to set up is a High Pass Filter around 60-80Hz. This will help you getting rid of wind rumble, footsteps and the 50/60Hz hum of an occasional ground loop. (However, do not use this when you're mixing sounds of explosions and sub-bass instruments.) Often you can set the gradient of the filter - the steeper the curve is, the more precise adjustment can be made in the frequency domain.

The next one should be a Low Shelf Filter with which you can emphasize the bass frequencies around 80-100Hz you've accidentally lost in the previous step. Another shelf filter should be a High Shelf Filter around 10-12kHz which you can brighten your mix with.

Between these two shelf filters you can always add several Normal Filters (or Bell Filters) to emphasize or suppress definite frequency bands. You can adjust their center frequency and Q factor (real mixing boards lack the latter) besides their level, so you can even cut a disturbing single frequency boost with them. Setting one around 400Hz and another around 2kHz makes a good starting point.

After all you'll want to save your settings to a brand new preset. You'll certainly find a Save or Save Preset button (often under the File menu or an icon of a disc) on the upper part of the plugin window, where you can give it a good name you will find next time as well (e.g. Mixing Board Preset).

Reducing Noise

Week 4: Reducing Noise -- Demonstrate how to reduce unwanted electrical and acoustical noise when recording.

It may be common sense, but avoiding noise during music production is of high priority. Most of the time you will want to make recordings in which only the instrument or the vocals can be heard. If you happen to be somewhere where the acoustics of the environment are worth recording (e.g. grand reverberations of a church, woodnotes and birdsongs of a forest), you should take care of recording it well -- otherwise, you'll want to get rid of echoes and background noises.

During pre-production you should make preparations, because with a few simple steps you can set up such a system which you can keep your recordings nice and clean with. First of all: turn every disturbing appliances off (such as television or air conditioner), be quiet and listen. Once you get used to the silence, you will find some quieter noise sources which can still be disturbing during a recording (fans of your computer, fridge in the kitchen, dogs barking outside, clocks ticking etc.). Move away from them if you can, or isolate the space which you are recording in. Turn the cooling of your computer down a bit (or clean the fans from dust), or even take it to another room; close the door to keep the noises of your household out; close the window, or simply choose a period of time when your neighbourhood is quite silent and you won't disturb them either (e.g. during working hours).

After you've taken care of acoustic noises, you also should take care of electrical ones. If you have the chance, use high quality equipment. Appliances and instruments of global brands are more expensive, but they tend to be of better quality and reliability. Shielded cables, reliable plugs and sockets, not the cheapest power supply units -- all these things can keep you away from electrical noise. There are less financially demanding methods as well: the shorter cables and the fewer pieces of gear you use, the less electrical noise you will find in your recordings. Turn off the dimmers and appliances you do not use, and use balanced cables wherever you can. Also try to avoid ground loop, which generates a disturbing hum of 50 or 60Hz (it depends on the power line frequency in your country).

After having your recording environment arranged carefully, you can start the production stage. If you happen to have a number of different types of microphones, you should consider using a directional one to isolate the instrument or vocalist from the environment. Directing a cardioid mic to the instrument and moving her closer to the sound source are excellent ways to have more direct sound and less background noise in your recordings. By this way you can also keep the signal level quite high, so you can keep the gain on a moderate level and thus keep electrical and background noises low.

During post-production you still have the chance to fix some noise issues, but you'll want to avoid that as much as you can, as they tend to be destructive or they consume way more time than proper preparations. Using a noise gate can keep background noises out, but by this way you'll lose all the lovely nuances such as the sound of the fingers sliding on the strings of an instrument. A long note fading out can also dissolve slowly in noise, and a gate can't do anything to save it. There are restoration effects such as de-noisers or de-hissers, but the processed audio does not sound perfect at all, since it loses its original timbre during the heavy filtering of the process. Prevention is better than cure.

Effects Basics

Week 3: Categories of Effects -- Teach the effect categories including which plugins go in each category and which property of sound each category relates to.

Using various effect plugins is a really fun thing to do, as you can alter the sound of any previously recorded audio to make it sound better, more exciting, or to suit any weird needs of your project. There are analogue and digital effects as well; here we're focusing on the digital ones, because nowadays Digital Signal Processing (DSP) is a thing any computer is capable of, but if you want to use analogue effects, you'll have to purchase them one by one, and these pieces of hardware can be pretty expensive.

The main purpose of the effects is the same: to alter the sound; but their methods can be really different. One can categorize the audio effects in many ways, e.g. like this:

• Dynamic Effects: they alter the amplitude of the audio. A Compressor (surprise!) compresses the sound, as it amplifies the loudest parts less than the others, so the volume will be more balanced, while the dynamics of the audio will be reduced. A Limiter is a really hard Compressor, as it doesn't even let the loudest parts exceed a given volume, however, heavy limiting can distort the sound. An Expander is the opposite of a Compressor, as it expands the dynamics of the audio making the quiet parts quieter and the loud parts louder. A Gate is a tool to mute ("lock out") some parts of the audio, so a Noise Gate can suppress the quieter parts completely (e.g. background noise) but won't do anything to the actual sound.
• Delay Effects: they alter the propagation of the audio. A Delay gives echoes to the audio, so any sound can appear again and again and again and again and again, causing feedback or fading out slowly. A Reverb puts the listener into a given sound environment with many echoes of different delay times and volumes and filters, so one can perceive the sound as being reflected from the walls and other surfaces of a bathroom, a hall, a church or even an arena. A Phaser, a Flanger or a Chorus also operates with delays and filters, producing sounds that are hard to describe but easy to recognize. (These three can also be categorized as Modulation Effects.)
• Filter Effects: they alter the timbre of the audio. A High Pass Filter lets the highs through unaltered but suppresses the lows, a Low Pass Filter makes the opposite, and a Band Pass Filter only lets through the sound between given frequencies (i.e. in a given frequency band). An Equalizer (EQ) lets you set your filters in any way you want to, using parameters (gain, frequency, Q factor, gradient, filter type) or a graphic interface (only gain for the given frequency bands).

I think one always has to hear an effect to find out what exactly it does, so I made a short audio clip using the following effects: Compressor, Delay, Reverb, Phaser, Flanger, Chorus, High Pass, Low Pass, Band Pass, respectively.

Recording Basics

Week 2: Recording Basics -- Recording audio in your DAW including preparing the project, creating the track(s), setting the click and countoff, and recording efficiently.

Preparing for your recording session is a really important part of music production, as it can determine how good or bad your final product will be. One of the most important things is rehearsal: even with the most versatile sound processing tools, one does not simply make outstanding music tracks out of sloppy, inaccurate performances (at best it will sound like Rebecca Black's Friday, where one can hear signs of excessive editing all along).

When you feel that you are ready to start recording, you will want to set your DAW up appropriately. I will use Magix Samplitude hereinafter, but I'm sure that the same settings can be found in every contemporary Digital Audio Workstation (DAW).

Fig.1: the Digital Audio Workstation

A piece of good advice can be to name your projects in a way that you will be able to understand even months later. Giving names like "Project01" or "New Project" won't lead you anywhere when you'll try to find one of your old recordings half a year later. So I advise you to come up with something like "MyBand - NewSong 2014" or "2014-07-31 Jamming with Tom". You will also want to put all your projects in their own folder inside your projects folder (e.g. D:\Samplitude work\Coursera Example), so you will always know where you can find the audio files.

Fig.2: giving your project a name, setting sample rate and number of tracks

In Samplitude, you can start a new project under File / New Virtual Project (VIP). Here you will be asked to choose the number of tracks (you can always add new ones later, if necessary) and sampling rate. In most home studio applications 44100Hz or 48000Hz should be enough. You should choose the former if you're going to burn your recordings on CD, and you should choose the latter if you want to use your recordings in videos. It is also possible to change the sampling rate later, but since resampling is destructive you should avoid it if you can (although most of the times a proper conversion won't make any audible differences).

Fig.3: setting audio device, buffer size and word length

The next step is to check the options of your audio device. Under File / Program Preferences / System/Audio you can find Audio System Options. There you can set the driver system (preferably ASIO, if you have a decent sound card or audio interface), the buffer size and the device resolution (bit depth, word length). The smaller the buffer size is, the shorter the delay between the input and the output will be, so you'll be able to monitor your input from your DAW. However, if you experience popping or crackling sounds during the recording, you should raise the buffer size and consider hardware monitoring. The default word length of a project is 16bit, but if you have appropriate sound files or a capable audio interface, you should choose 24bit. It gives more resolution to your audio data, more headroom to your recordings, signal processing can be more precise, and in the end you can still export your project in 44100Hz/16bit for audio CDs and mp3 conversion (or you can leave it at 48000Hz/24bit for DVD applications etc.).

Fig.4: recording options

Under Play/Rec / Record Options you can set the recording format (Wave, uncompressed), word length of the sound recording, monitoring mode, name of takes etc. If you go back to System Options (or click the corresponding button on the main screen), you can find the click or metronome options. You'll want to set the metronome to be active while recording but inactive during playback, and setting one or two measures of precount can also be useful to catch the beat when the recording starts. You can set the click sounds here, and on the main screen you can set the Beats Per Minute (BPM) to set the tempo (default is 120) and the measures (bars) to be used (usually 4/4, but other ones like 3/4 or 12/8 also may be used).

Fig.5: metronome options

Finally, you should name your track you want to record to, set it to mono if you're using a single microphone or instruments like an electric guitar, or set it to stereo for a pair of microphones, a stereo mic or a stereo instrument (e.g. a synthetizer). Set your audio interface up (plug the cables in, switch phantom power on if necessary, set the gain), set the track in the DAW armed to record using the button with the red dot (next to the Solo and Mute buttons), and then you can start recording by pressing the Record button down below.

Fig.6: finally, ready for recording

One more thing: you should consider using a decent pair of headphones to monitor what you're recording and the backing tracks. This will elliminate feedback, the backing tracks won't be recorded again by the mic, and last but not least you will be able to live in peace with your neighbours :)

Microphone Basics

Week 1: Microphone Basics -- Type, Frequency Response, and Polar Pattern.

Microphones are transducers which convert acoustic energy into electric current. The sound source (e.g. a vocalist or an acoustic instrument) makes the air vibrate, and these waves of compressions and rarefactions propagate in the medium. When these waves reach our ear, we hear them as sound. A microphone works quite like a human ear: the sound waves pass through the grille and the windscreen of the mic (~outer ear); reach the membrane (~ear drum) and set it in motion (~middle ear); the motion of the membrane is converted into voltage/current variations (~inner ear) by a permanent magnet; these variations pass through the microphone cable (~nerves) and finally they are processed by the audio interface (~brain).

There are several types of microphones.

• Dynamic microphone is quite a solid type, as she can take the extreme conditions of being used on stage with ease. Should an exhibitionist vocalist grab her by the cable, swing her above his head, drop her accidentally, or even pour some beer on her, a dynamic mic may handle all this impoliteness well. Also, she provides a nice, warm sound suitable for vocals, as well as bass and middle oriented instruments. In other words, her frequency response isn't flat: she tends to emphasize frequencies between ~2kHz and ~8kHz, and suppress trebles above ~10kHz. Bass can also be suppressed a bit (below ~100Hz) -- however, it can be (over)compensated by the proximity effect.
• Condenser microphones are more suitable for use in professional and home studios, as they tend to have a more natural and crisp sound. This type is more demanding than a dynamic one, as she needs careful handling and also phantom power (48V) from the audio interface (or for some types, from a battery). Condenser microphones are more sensitive; with this type you can record more details of the sound environment, i.e. reverberations of the room, but background noises as well. She is better at recording sound sources with a lots of treble (percussions, cymbals, brass instruments, etc.) on stage, but she can be used to record vocals or acoustic guitars as well -- or almost everything in a studio. The frequency response of a condenser mic is pretty flat in general, so she "hears" bass and treble equally well, and there are hardly any colorations in the sound. This makes her recommended to studio quality recordings -- however, a dynamic mic of good quality can provide a more live or "hot" vocal sound.
• One can also meet ribbon microphones and piezo microphones, but they are far less widespread in studios or on stage. The former ones can be rather vulnerable but are said to sound great on vocals, while the latter ones are used mostly in mobile phones as they are small and cheap to produce, although their sound quality is questionable. However, piezo mics are used as contact microphones in acoustic guitars quite often, as they can take vibrations of solid bodies of instruments well.
• Fun fact: a dynamic microphone works just like a reversed speaker. If you sing loud enough into a speaker or a set of headphones, you might be able to record your voice with it when the system is assembled accordingly. This method is becoming more and more popular being used in front of kick drums, as a large speaker (woofer) can record bass well and suppress treble coming from the cymbals at the same time. (You can try this in small when connecting two pairs of earphones to a Y-adapter, wearing one pair and tapping the earpieces of the other (or simply letting them collide).) You can also listen to music via a dynamic mic, although there are way better methods to do this (and it can also be pretty dangerous for the mic).

Both the dynamic and condenser microphones can have different sensitivity in different directions. It means that a particular mic can "hear" perfectly in one direction and worse in another. This direction dependent sensitivity is plotted on the polar pattern (see the figure below), and this can define the possible applications of the microphone.

Fig.1: Polar Patterns in 2D
(source: https://jwoodardvt.wikispaces.com/file/view/polar_patterns.jpg/425698960/polar_patterns.jpg)

As you can see, there are circles, buns, upside down hearts, double buns, dragonflies or just funny blobs or spots on the figure. Actually, they represent how sensitive the microphone is in a particular direction, when the main axis of the mic is pointing upwards. The more the spot approaches the outer circle (dotted line), the more the microphone hears in that direction. Maybe the following figure can help a little:

Fig.2: Polar Patterns explained in 3D
Note: the labels "Cardioid" and "Supercardioid" had been switched accidentally!
(source: http://soundtech.files.wordpress.com/2009/02/mics-polar-patterns0101.png)

Microphones of different polar patterns can be used in different situations.

• An omnidirectional mic hears in every direction around herself, so she can be useful when recording the sound of the whole room, all the reverberation, or even several speakers or vocalists around the mic.
• A cardioid mic hears well from the front (on axis) but hears almost nothing from the back. She can be useful when one wants to record only a single vocalist or instrument and not the others (also not too much of the reverberation), and on stage the vocalist can turn the rear of the mic towards his stage monitor to prevent feedback. Proximity effect is also to be considered, as when a cardioid mic is close to a speaker or the vocalist's mouth, its bass sensitivity rises sharply, and one can use that to their advantage.
• Supercardioid and hypercardioid mics are quite the same as the cardioid ones, but they hear a bit more from the back.
• A bi-directional mic hears well from the front and from the back, but not from the sides.
• A shotgun microphone hears almost only on axis, so it can be useful to record a particular speaker or sound source from afar.

That's all for now, we will continue next week :)

Introduction

Hi there.

I am Boti from Budapest, Hungary and this is my first blog post about music production. All this is a part of the Peer Review Assignments in the course Introduction to Music Production on Coursera. I am currently trying to get prepared for a language exam in English, so I'll try and do my best here.