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

Introduction
We came to the Final Week of the Course and I must said that Loudon Stearns did a good job to introduce us to the various aspect of Music Production. It is kinda a bit sad to say goodbye now as there are definitely more stuffs that we can learn. And definitely I will be getting my hands on some gear to do some editing myself.

The topic I choose this week is "Explain the usage of the 5 most important synthesis modules: Oscillator, Filter, Amplifier, Envelope, and LFO" and as before, I will be using material from the course lectures.

Overview
The 5 important synthesis modules are the Oscillator, Filter, Amplifier, LFO and the Envelope. And they are depicted in the picture below 

Overview

Oscillator

The oscillator, otherwise known as the voltage controlled oscillator, generates sounds based on geometric waveforms. The sounds its generates gets it by the shape as shown below : 

Waveform

For instances, a square wave is squarish. Examples of sound generated by a square wave includes the clarinet. The sawtooth waveform is a sawtooth. Example includes a guitar where the guitar string is plucked and released. The important aspect is that these shapes gets modulated over time and hence their pitch and frequency changes over time. 

Oscillator

The diagram above shows how the oscillator would change over time via a sawtooth waveform. 

Filter

Filters are happening everyday in our lives. An example of filter is the throat as your vocal and the mouth as the filter. As you change the shape of your mouth, the sound sounds different even though the sound is produced at the same pitch and frequency.

There are mainly 4 types of filter in the synthesizer. The low pass filter, the high pass filter, the band pass filter and the band stop filter.

Low Pass Filter

The low pass filter basically allows the low end of the sound to pass. There will be a cut off frequency at which the sound level will drastically decreases.

Low Pass Filter With Resonance

Low Pass Filter can be amended to add resonance. The resonance emphasis the harmonics of the sound. 

High Pass Filter

The high pass filter cuts off the low ends of the sound. This is very useful in eliminating surrounding noises such as your CPU fan, your breathing etc.

 Band Pass Filter

The band pass filter is useful in emulating the frequency of the voice. 

Band Stop Filter

The Band Stop Filter generally creates a notch hollow frequency. 

Amplifier

The third part of the filter is the amplifier, otherwise known as the voltage controlled amplifier. This is defined by the envelope show here : 

Envelope

The envelope is generally defined by 4 things, the ADSR, or The Attack, The Decay, The Sustain and The Release.

The Attack, or Attack Time is the time the sound gets from 0 to 100% within the specific timeframe. 

The Decay Time is the time the sound gets from 100% to the sustain level within the specific timeframe.

The Sustain Level is the level at which the sound is at a steady state.

The Release Time is the time the sounds stays from the sustain level to 0 within the specific timeframe.

The Amplitude Envelope

There are several types of different amplitude envelope as shown below :

 

Low Frequency Oscillator 

The Low Frequency Oscillator is a modulator that is cyclic in nature. It is termed low frequency as the frequency it uses is usually lower than the range of the human hearing. It is used to control the destination such as an oscillator and is useful to mimic vibratos. In every LFO, there are 3 parameters that will be used. 1) The Source, 2) The Destination, 3) The amount of modulation.

An FLO in action

The picture above shows a triangle waveform as the source, with the oscillator as the destination and the amount of modulation at 1.21Hz.

Reflections

It most probably has been a tough 6 weeks for me as I am a working adult and there are lots of works and everyday commitment that I do while I am working. But I am glad I have pushed through and made it here. Although there is still the Final Exams. I hope you have enjoyed the course as well and learn somethings along the way. Thank you for reading through and I do apologise for my grammer. It is not perfect. 

Wish You Well In Your Musical Journey!

Demonstrate the configuring of an EQ plugin to function like a large format mixing console EQ section. You can use the settings shown in the material or base your settings off of the manual of another mixing board. Include instructions showing how to save the setting as a preset in your DAW.

Introduction

Hey there, it has really been a hectic 5 weeks and here we are. The course would be over very soon. It has been nice taking this course and it surely has been quite informative for me! This week, I will be touching on how you can configure a digital EQ to mimic the real large format mixing board. I will be using materials for the lectures.

The Contemporary Large Mixing Board EQ

The figure above shows the contemporary large mixing board EQ. It comprises of :

• The High Pass Filter
• The High Shelfing Filter
• The High to Mid Range EQ
• The Low to Mid Range EQ
• The Low Shelfing Filter.

It is important to mimic a real mixing board as it is much easier to add something to the digital copy then to do it via the real hard ware.

The High Pass Filter

The high pass filter shows here is designed to remove any rumble or noise. Anything below the frequency will be cut off. Here, anything below 75Hz will be cut off. The steepness of the slope is -18dB/OCT but it is set as -24dB/OCT in the digital EQ

The Low Shelfing Filter

The low shelfing filter here will give weight to the bass or warm to keyboards. It is set at 80Hz and +/- 15dB. It adds quite a bit of bottom end bump as shown in the picture below :

The Low to Mid Range EQ

The Low to Mid Range EQ are used for frequency from 100Hz to 2000Hz. The gain can also be adjusted to +/- 15dB. However, in real formatting, it is not recommended to use such large gain settings. Usually the maximum gain to use is +6dB, 

The High to Mid Range EQ

The High to Mid Range EQ is used for frequency from 400Hz to 8000Hz. The gain again here can be adjusted to +/- 15dB. But again, it is not recommended to use such large gain values. 

 The High Shelfing Filter

 

The High Shelfing Filter is used for frequencies of 12000Hz. Again, it has a gain of +/-15dB.

Finished Product

The finished product is shown here and all is set at the default zero value first before any mixing occurs. Do kindly save it as the default EQ setting so that you can access them quickly again.

Saving In Sony Sound Forge

For Sony Sound Forge, it is recommended to create a new folder under the FX Favourites menu and fill it with the most commonly used EQ plug-ins that you used. First, right-click an empty spot and create a new folder. Name it to your liking. In the tree list, Click All to view all of the plug-ins loaded into the system in the list view. Select The EQ that you want to add into the folder and add them to the selection.

Finally, drag any of the selected items and drop it into the folder that you have just created. They will now appear in your created folder. Do take note that they will still appear under the All folder as well.

These EQs will also be easily found under My Favorites menu. under your My Favorites menu.
(References : 10 editing tips to help you work faster in Sound Forge Pro)

Reflections
As mentioned earlier, it sure has been quite a lot of information these 5 weeks. I am actually quite glad I pick up this course because I am learning so many things that will help me later on. It is also fascinating to know that there is so much more to contemporary music and music editing etc. It surely makes you appreciate the efforts musicians puts in to create their own music album! I won't be a master in 6 weeks time, but I am sure I learn something. 

Explain Dynamic Range and the many ways producers manipulate dynamic range.

Introduction
Music has also been a comfort and pleasure zone for me and the stresses of life. I am pursuing this course to know more about music.This week, I am picking the topic on Dynamic Range, its explanation and how producers manipulate the dynamic range.  I have taken pictures out of the videos of the course to explain some of the terms here.

Dynamic Range
Performer
To a performer, the dynamic range is the musical control of volume over the song; or how to play the score. It determines how a performer plays the music, for instances, more aggressive playing at the chorus or solo part of a music.

For example, p or piano, meaning 'soft' or f or forte, meaning 'loud'. mp means moderately soft and mf means moderately loud

More information on musical dynamic range can be found here.

Musical Equipment
For a piece of musical equipment, the dynamic range is the range at which the instrument or equipment can reproduce accurately and properly. This range is usually between the noise floor, the quietest level at which the equipment can reproduce. And the loudest level, also known as the distortion

 Human Ear
For the ordinary human, the dynamic range is more complicated as the human brain acts as a automatic gain control for the range that we can hear. In very quiet place, it is possible to hear sound like your own heartbeat while at noisy places such as live band performances, the quiet noise gets cancelled out and you only hear the loud noises. 


The measurements of the dynamic range of a song is usually defined as the amplitude, measured in dBSPL or decibels in relations to sound pressure level. 0 is the minimum. This is the quietest noise that the normal human can hear. This is also known as the threshold of hearing. The maximum noise a human can hear, also known as the threshold of pain is between 120-140 dBSPL.

Dynamic Range Manipulation

On a macro scale, dynamic range manipulation is done via using the Volume Fader & Volume Automation. This is done by the adjusting the relative section between the levels. Here, the chorus can be increased by a few decibels and decreasing the verse by a few decibels. 

On a smaller scale, like the dynamic range of a single performance. For instances, during singing, The Singer will sing louder during the chorus and quieter during the verse. For a song, the singing is the focus and hence it is the mix engineer jobs to keep a clear focus for the listener.

This technique, known as "Riding the Fader" is a kind of manual compression. This keeps the focus on the vocalist performance.

On the micro-scale, the dynamic shape of each individual drum hit can be controlled.This is usually done via compressors and gates. Here, each transient is manipulated automatically. Transient means the place where the amplitude changes a lot in a short period of time. Examples includes clapping and drum hitting.

A Transient

As the changes is too fast, it is usually done via programming or algorithms controls. 

In Dynamic Manipulation, it is important to be true to the original musical content. This is to let the listener focus on the musical instrument at the moment in time of the music piece. 

Dynamic Range Terms
There are usually 4 common terms used in terms of dynamic range manipulation and these are 1) Compression, 2) Expansion, 3) Limiters and 4) Gates. 
 
Compression
In dynamic range manipulation, it is common to use compression where the dynamic range is reduced. This is done by making the loud quieter or the quieter louder. 

Expansion
Expansion is done where the dynamic range is emphasized. Here the loud is made louder and quieter quieter.

Gates
Gates, or more commonly known as noise gate is traditionally used to remove the hiss and noise of a piece of musical instrument playing. For instances, when a guitar gets playing, it is usually louder than the ambient noise. But when it stops, the hiss or noise comes on. Setting a gate where the threshold is above the noise but below the signal of the guitar effectively removes these noise.

Limiters
Limiters is a type of extreme downward compression where the compression ratio is more than 10:1. It is traditionally used as a form of protection. For instances, when a vocalist screams or drops the microphone. The limiters will set in and cut off the sound.

In contemporary music, it is used mainly as a loudness maximizer. Here, the sound is made to be as loud as possible without actually increasing the amplitude. This is done by changing the timbre of the songs.

Reflections
I have gotten hold of Sony Sound Forge Audio Studio 2014 and will be raring to have a go at what I have been taught. Unfortunately, my laptop, which is in its 4th year is starting to crank up on me. So most probably I need to get a new piece of gear before I can get cracking.

Thank you for taking your valuable time to read this and hope to see you next week!
 

The Channel Strip

Introduction

This week assignment, I am taking on the channel strip. However, for the channel strip, I will be using the 5088 Channel Strip by Rupert Neve Designs instead of the one shown by Loudon Stearns.

As always, the channel strip is module and each channel repeats itself. The sounds or signals flows from top to bottom.

A Channel Strip

Input Select

Push buttons at the top are provided to select the transformer coupled line, Portico buss and tape inputs. There is also a phase reverse push button. This rotates the phase by 180 deg at the input.

Input Trim

The input trim provides adjustment of the gains of the selected input by +/- 10dB.

Group Send Select

9 push buttons provide the buss assignment of the channel input signal to the 8 subgroup modules and the stereo Buss. 

Aux

Aux 7/8 sends the signal to the pre-fader and before the soft mute, rather than the default post-fader.

Aux 5/6 offers dual mono or stereo operation. The level 6 controls operates as a pan while the aux 5 pot controls the stereo level. When MONO is pressed, 6 becomes a separete level control. The Mute switch which mutes Aux 5/6 and Aux to Group when engaged, routes 5/6 to the selected subgroup busses.

Aux 3/4 and Aux 1/2 are identical pairs. It has a center detent and serves as a pan or level control. As before, there is a mute and pre/post fader switch. There is also a SFP which is useful for creating cue and stereo reverb sends.  

Channel

The Channel Pan send the signal to eigher the left or right side of the stereo. 

The Channel Mute mute the entire channel while the Channel stereo is useful for isolating the track for solo play.

Reflections
There are a lot of topics this week so you guys better stay attention while i play catch up! Cya next week!

The Analog to Digital conversion process.

Introduction
The analog to digital conversion process is actually the conversion of the pressure variations in the air to binary information that the computer can process. This is done via the process known as sampling

Fig 1. Analog to Digital to Analog Process (Source : Wikipedia)

Sound
Sound is actually pressure variations in the air. It is vibration that propagates as audible mechanical wave of pressure and displacement. Sound needs a medium to travel and it is usually air most of the time. 

Binary Information
However, a computer cannot process such information. For a computer or DAW (Digital Audio Workstation) to process this information, it would require the information to be presented in bits. A bit is simple a "1" or a "0". 

1 bit is simply 2 wordlength, 2 bit is 2^2, which is 4 wordlength. A simple permutation of the bit will be as below 

1 Bit = 2^1 = 2

2 Bit = 2^2 = 4

3 Bit = 2^3 = 8

4 Bit = 2^4 = 16

5 Bit = 2^5 = 32

6 Bit = 2^6 = 64

7 Bit = 2^7 = 128

8 Bit = 2^8 = 256

9 Bit = 2^9 = 512

10 Bit = 2^10 = 1024

11 Bit = 2^11 = 2048

12 Bit = 2^12 = 4096

13 Bit = 2^13 = 8192

14 Bit = 2^14 = 16384

15 Bit = 2^15 = 32768

16 Bit = 2^16 = 65536

Sound In Digital CD

CD audio has a sampling rate of 44100Hz and 16 bit resolution for each stereo channel. Analog signals that have not been already been bandlimited must passed through anti-aliasing filter before conversion. This will prevent distortion caused by audio signals with frequencies higher than the Nyquist frequency. The Nyquist frequency is half of the system's sampling rate. For CD, the Nyquist frequency is 22050Hz which is actually the upper range of the human ear hearing limits. Hence, a CD is able to record exactly what a normal human can hear. 

Sampling

The conversion from analog to digital, otherwise know as the sampling process depends mainly on 2 things. The sampling rate and the dynamic range, otherwise known as the resolution.

It is common in recording to use sampling rate of 48,000Hz. This is higher than the CD sampling rate.

A complete list of audio sampling rate is as below : 

  

Sampling rate	Use
8,000 Hz	Telephone and encrypted walkie-talkie, wireless intercom[10][11] and wireless microphone[12] transmission; adequate for human speech but without sibilance; esssounds like eff (/s/, /f/).
11,025 Hz	One quarter the sampling rate of audio CDs; used for lower-quality PCM, MPEG audio and for audio analysis of subwoofer bandpasses.[citation needed]
16,000 Hz	Wideband frequency extension over standard telephone narrowband 8,000 Hz. Used in most modern VoIP and VVoIP communication products.[13]
22,050 Hz	One half the sampling rate of audio CDs; used for lower-quality PCM and MPEG audio and for audio analysis of low frequency energy. Suitable for digitizing early 20th century audio formats such as 78s.[14]
32,000 Hz	miniDV digital video camcorder, video tapes with extra channels of audio (e.g. DVCAM with 4 Channels of audio), DAT (LP mode), Germany's Digitales Satellitenradio, NICAM digital audio, used alongside analogue television sound in some countries. High-quality digital wireless microphones.[15] Suitable for digitizing FM radio.[citation needed]
44,056 Hz	Used by digital audio locked to NTSC color video signals (245 lines by 3 samples by 59.94 fields per second = 29.97 frames per second).
44,100 Hz	Audio CD, also most commonly used with MPEG-1 audio (VCD, SVCD, MP3). Originally chosen by Sony because it could be recorded on modified video equipment running at either 25 frames per second (PAL) or 30 frame/s (using an NTSC monochrome video recorder) and cover the 20 kHz bandwidth thought necessary to match professional analog recording equipment of the time. A PCM adaptor would fit digital audio samples into the analog video channel of, for example, PAL video tapes using 588 lines by 3 samples by 25 frames per second.
47,250 Hz	world's first commercial PCM sound recorder by Nippon Columbia (Denon)
48,000 Hz	The standard audio sampling rate used by professional digital video equipment such as tape recorders, video servers, vision mixers and so on. This rate was chosen because it could deliver a 22 kHz frequency response and work with 29.97 frames per second NTSC video - as well as 25 frame/s, 30 frame/s and 24 frame/s systems. With 29.97 frame/s systems it is necessary to handle 1601.6 audio samples per frame delivering an integer number of audio samples only every fifth video frame.[9]  Also used for sound with consumer video formats like DV, digital TV, DVD, and films. The professional Serial Digital Interface (SDI) and High-definition Serial Digital Interface (HD-SDI) used to connect broadcast television equipment together uses this audio sampling frequency. Most professional audio gear uses 48 kHz sampling, including mixing consoles, and digital recording devices.
50,000 Hz	First commercial digital audio recorders from the late 70s from 3M and Soundstream.
50,400 Hz	Sampling rate used by the Mitsubishi X-80 digital audio recorder.
88,200 Hz	Sampling rate used by some professional recording equipment when the destination is CD (multiples of 44,100 Hz). Some pro audio gear uses (or is able to select) 88.2 kHz sampling, including mixers, EQs, compressors, reverb, crossovers and recording devices.
96,000 Hz	DVD-Audio, some LPCM DVD tracks, BD-ROM (Blu-ray Disc) audio tracks, HD DVD (High-Definition DVD) audio tracks. Some professional recording and production equipment is able to select 96 kHz sampling. This sampling frequency is twice the 48 kHz standard commonly used with audio on professional equipment.
176,400 Hz	Sampling rate used by HDCD recorders and other professional applications for CD production.
192,000 Hz	DVD-Audio, some LPCM DVD tracks, BD-ROM (Blu-ray Disc) audio tracks, and HD DVD (High-Definition DVD) audio tracks, High-Definition audio recording devices and audio editing software. This sampling frequency is four times the 48 kHz standard commonly used with audio on professional video equipment.
352,800 Hz	Digital eXtreme Definition, used for recording and editing Super Audio CDs, as 1-bit DSD is not suited for editing. Eight times the frequency of 44.1 kHz.
2,822,400 Hz	SACD, 1-bit delta-sigma modulation process known as Direct Stream Digital, co-developed by Sony and Philips.
5,644,800 Hz	Double-Rate DSD, 1-bit Direct Stream Digital at 2x the rate of the SACD. Used in some professional DSD recorders.

 Table 1 : Sampling Rate (Source : Wikipedia) 

The other component of sampling is audio resolution, this is dependent on the wordlength or bit depth of the sample. It is common in recording to use 24wordlength sample; which is 2^24 = 16777216. This is done via using pulse-code modulation (PCM). 

Variations in bit depth affects the noise level from quantization error. 

In simple terms, increase in sampling rate will present the audio file more accurately.

Reflections
I am not that familiar with DAW hence, I have choose this topic. Definitely I am getting a DAW. Knowing the Analog to Digital Conversion process will defintely help in using a DAW more efficiently. I hope you have enjoyed reading these information. Cya Next Week!