Creating an Additive Synthesiser in Synthmaker

Additive Synthesiser           

The following, is a quick post on the methods used to create an additive synthesizer within the software program Synthmaker.

This software was available for free with issues of computer music magazine,the software goes by a different name nowadays but any modular synthesiser program has the same principles described here.

This software is similar to programs such as reactor so many of the principles can be applied to whatever synthesiser building program you may be using.

 The report will highlight the methods used to create the synth, the intentions behind creating the synth and how successful the end product was.

Additive Synthesis

            The principles of additive synthesis are fairly simple and rely on basic sine waves to provide the basis for the sounds which will be created by the synthesizer. 

The basic premise of additive synthesis, is that a waveform can be described in terms of the frequencies and amplitudes of those frequencies contained within the waveform. 

This means that a certain amount of sine waves can be blended together at particular frequencies and amplitudes can be used to create different waveforms such as square or saw tooth waveforms.

 The character of the waveform, will be dependent on the harmonic intervals determined from the fundamental frequency. 

A very basic form of additive synthesis is the Hammond organ, where, drawbars on the organ are used to add higher harmonics as they are pulled out , creating a change in the character of the sound. In creating the vst plug in, the example of the organ can be used as a basis to start the design of the plug in.

            The first stage, involves using a combination of sine waves to produce a fundamental frequency and the harmonic information that the fundamental would produce in a real world instrument. At least 9 individual sine waves would be employed within the additive synth to create some harmonics at differing amplitudes to provide some character to the basic synth sound. 

The basic additive synth was used as the blueprint for the synth but some extra harmonics would be added to give more harmonic depth to the sound. An 8^th and 9^th harmonic were added into the additive module, adding some further high frequency content to the overall tone. 

The harmonic values of 8 and 9 were typed into the float integers and amplitude values of 0.2 were set, initially, for both harmonics. 

The amplitude values of each harmonic note could be altered individually or via a controller which could provide increased possibilities to alter the sound in real time. The sounds at this point would vary from high frequency tones to low bass tones depending on the amplitude values set for each harmonic. 

There are still many factors that need further consideration if the additive synthesizer is going to have more variation of sounds.

            Time and how a sound changes over time is a key factor in a sound being interesting to listen to.

 In a stringed instrument, the natural inclination of the sound is to be bright at the initial attack of the sound but over time, the higher frequencies fade away more quickly than the lower frequencies. 

This can be replicated within the synth by controlling the low and high frequency harmonics separately, using an envelope. 

This was done by creating a separate output for the 1^st to 5^th harmonics from the additive and then creating a separate output for the 6^th  to 9^th harmonics from the additive module.

These separate outputs were then connected to an ADSR envelope module separately

Doing this , allowed some control over how the harmonics would behave and the low and high frequency content could be controlled to react differently over time from each other.

 The intention, is to simulate the behavior of a stringed instrument where the high frequency content will fade more quickly over time than the low frequency content. 
This is a starting point for the synthesizer but more interesting results may be achieved by adding some control to the individual harmonic components themselves.

 The ADSR envelope was set to a particular value and then made into a module in order to keep the internal synthesizer components in order. Each individual harmonic or particle of the additive synthesizer would have a separate slider control which would determine the amplitude value of each separate harmonic, therefore, allowing a greater degree of control over the overall sound. 

This was done by connecting a vertical slider to each harmonic’s amplitude value whereby each slider could then be controlled on the synthesizer front panel.

            The intention was to provide a similar idea to the drawbar system on a Hammond organ, where pulling out the draw bars on the organ will add more harmonic content to the sound.

Some modifications were also made inside some of the vertical sliders where the value scale of the vertical slider would be increased from 0 to 1 up to 0 to 10.

 This was designed to give extended control to the amplitude values of each harmonic with increased possibilities in changing the sound of the plug in.

            As part of the process, each vertical slider’s scale would be altered differently in order to give a specific character to each harmonic. 

The slider would then be given a label/name describing the characteristic of the sound which would then be clear on the front panel, giving the user a preset description of each vertical slider on the front panel. 

The intention was to provide a wide range of sounds through adding different harmonic components at varying amplitudes. 

The synthesizer would be routed in two different sections. One section would feature the higher harmonics whilst the second section concentrated on the low to mid range content.

 Each section also had its own envelope, allowing control over the sustain of the frequency content within each section. This could enable the user to tailor the envelope controls of each section whereby the higher frequencies tail off quickly in a manner evoking the characteristics of stringed instruments where much of the high frequency information exists in the attack portion of the envelope rather than in the release of the note.  

            After the additive and ADSR  envelopes, a further reverb module was added to give a sense of space to the basic sounds being produced from the harmonics. The reverb unit was customized from an existing unit but new knobs and their values were applied to the float integers to create different values for the room size, width and effect mix. These values were set by drilling into the individual knob controls and setting parameter values.

This allowed for some fairly subtle reverb settings which would provide some depth to the overall synthesizer sound.

Volume and Strength Controls

            After the reverb module, the volume control module was added which would produce a non linear behaviors for the action of the volume and strength control knobs connected to the bender module and situated on the front panel as volume and strength controls. 

The volume and strength controls were connected to two float integers which would then connect to the bender module, determining the non linear control of the volume and strength controls which would also be visually represented by a level meter. 

The volume and strength parameters were also adjusted by drilling into each of these respective knob controls and setting new levels for the minimum and maximum settings.

            To provide some quick settings for users, a preset manager was integrated onto the front panel and some quick preset files and sounds were created and appropriately named for quick use.

 Four presets were named, Rhodes, Rhodes reverb, steel drums and organ. The vertical sliders within the additive module were untidy with a clumsy layout so these were all converted into modules, attached to the float integers, which were also converted to modules, and connected to the harmonic components.

Harmonic sliders within the additive tidied by converting into modules

The final internal working of the synthesizer can be seen in the screen shot below

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Equinox Sounds: MIDI Loops, WAV Sample Packs & VSTi Sounds Ambient Chillscapes Vol 2

Equinox Sounds: MIDI Loops, WAV Sample Packs & VSTi Sounds Ambient Chillscapes Vol 2



This pack is the 2nd volume of ambient chillscapes created by papas st germain for Equinox sounds.Inspired by artists such as boards of canada,brian Eno,Harold budd and burial.Designed to be used in any genre.

Equinox Sounds: MIDI Loops, WAV Sample Packs & VSTi Sounds Ambient Chillscapes Vol 2

Equinox Sounds: MIDI Loops, WAV Sample Packs & VSTi Sounds Ambient Chillscapes Vol 2



Papas st germain's latest release for equinox sounds

Downtempo Chilltrax Vol 1 by Papas St Germain at Sampleism

Downtempo Chilltrax Vol 1 by Papas St Germain at Sampleism



This is the brand new sample pack from papas st germain.The pack will start at an introductory offer of 50 % off along with offers on all of papas st germain's other sample packs.

AMBISONIC SOUND  History and Future By PAPAS ST GERMAIN

AMBISONICS : THE SECOND COMING 

This post, is an investigative report on Ambisonic surround sound and its place in modern music technology. The report will investigate the history of Ambisonics, current developments in Ambisonics and potential future developments in this area in conjunction with future progress in music technology.

 This post will attempt to highlight some of the advantages of Ambisonic surround sound over its competing surround technologies and why it has the potential to still be relevant 40 years in to its existence.

Many of the ideas covered in the following post, relate to surround sound technology and its development since its inception, in varying forms, in the early 1970s.

 Ambisonics emerged as one of these surround formats and quickly appeared to have some advantages over other surround sound developments such as quadraphonic sound.

 Ambisonic surround was created in the early 1970s and there were several key players in helping to create this technology.

 Michael Gerzon was particularly prominent in the research and development of Ambisonics along with various cohorts, professor Felgett, John Hayes, John Wright and Dave Brown. The groups work was based at Reading University and was well received and respected within the audio community, gaining backing from the National Research and Development Corporation.

Research performed by this group at Reading University resulted in many different patents connected to the pioneering work undertaken in developing Ambisonic systems.

Ambisonic technology became sidelined, largely due to frustrations with bureaucracy and an inability to gain a deal with Dolby Labs in the USA.Much of the pioneering work in Ambisonics can be referred to at websites such as Ambisonics.net, where many of the papers proposed by Michael Gerzon and company can be referenced.

Other sites such as Ambisonic.info,audiosignal.co.uk and blueripplesound.com are useful in investigating the history, current state and future developments of Ambisonics whilst articles in sound on sound magazine(August 2001, You Are Surrounded) are useful for gaining some insight into the technology and how it could potentially be applied to modern music technology.

The basic theoretical principles of Ambisonics will be studied, highlighting the advantages of the Ambisonic system, both in terms of recording and reproduction and the areas within the audio industry where Ambisonic surround could be implemented in favour of other surround sound formats.

 The advantages of Ambisonics over surround sound formats such as quadraphonic sound and 5.1 surround will be examined as well as the flexibility of the system in terms of integration with other formats.

The final area of the post will attempt to address the future developments in Ambisonics and how those developments will enable the system to be relevant to modern music technology within the wider entertainment industry.

Ambisonics : The Basics

The key intention of Ambisonic recording, is to capture sound from all around equally from a single point. This is physically impossible but in their pioneering research, Gerzon and Felgett used complex maths to calculate what could be heard at a minute single point in the centre of the sphere.

 Gerzon and Felgetts work had its roots in Alan Blumlein’s coincident stereo techniques of the 1930s but expanded on Blumlein’s work, implementing maths and psychoacoustics to arrive at the Ambisonic sound system.

 In simple terms, Ambisonics is a periphonic sound recording, synthesis and playback system. An Ambisonic system can record and playback sounds from left and right, front and back and also up and down.

 These features make Ambisonic sound a genuine 3 dimensional sound proposition. Ambisonic technology breaks each directional part of the sound field into separate spherical harmonic components named W,X,Y and Z.

 W is taken as the amplitudinal reference point for the overall signal whilst X,Y and Z represent the directional axis, Z representing the vertical axis. Part of the theory of Ambisonics, involves putting each of the directional signals W,X,Y,Z through shelf filters, each of which, have different gains at low and high frequencies.

These are intended to replicate the manner in which the human ear and brain pin point sound, helping to determine its direction and distance.

In recording terms, there are different techniques in the manner in which Ambisonic sound can be captured. The Nimbus – Halliday array uses 3 microphones which are positioned as near coincident as possible.

Two of the microphones will have figure of 8 capsule settings which will provide the front-back and left-right signals whilst the remaining microphone will have an omni setting, comprising the W component which references the amplitude information affecting the overall array.

 The signals recorded from this array can be reproduced in stereo or mono whist a decoder can be used to implement a horizontal image. The recording alternative to the Nimbus Halliday array, is the actual Ambisonic microphone known as the sound field microphone.This microphone comprises of 4 cardioid capsules which are configured in such a way that they form a regular tetrahedron shape.

 The microphones outputs are referred to as A format signals but further processing is required (through a decoder) to derive signals compatible with the standard B format of Ambisonic systems.

 The Soundfield microphone has its capsules set up in this configuration in order to capture a physical representation of a spherical pick up with the least amount of pick ups possible.

 This doesn’t entirely solve the problem of capturing as accurate a representation of the sound field as possible. There are still time differences between each capsule as they are physically separate from each other.

The aim is to replicate the signal that would be captured at a single point in the middle of the array. By performing the correction to these time differences, the B format signals should be accurate in the time domain up to about 15Khz.

 The Sound field microphone can correct the B format signals as close to a flat frequency response as possible for sounds arriving at all directions.

The advantage of this, is that the angle of incidence by which a sound arrives at each capsule will not have a noticeable effect on the sound captured. A recent example of this microphone being used in practice can be seen in a recent recording by producer Mark Hornsby.

 Hornsby used the Sound field SPS200 multi capsule microphone for a recent orchestral performance at Abbey Road Studios. The intention was to capture the soundspace in 3 dimensions to be replayed in the 4 channel format. The de coding process is handled by the Soundfield sps 200 surround zone plug in.

 This plug in enables the recorded audio to be reproduced in various different formats. The key factor in using this system for Hornsby was the quality, flexibility and portability of the system. “ I think of it more as a recording system than just a microphone. Having the ability to re point microphones in the mixing session after the session is amazing to me”.

This quote from Hornsby highlight the usefulness of an Ambisonic system where one microphone and a software encoder allows for a portable recording system.

Historic Development of Competing Technologies

Surround sound technologies have been in existence in various forms throughout audio recording history with varying degrees of success. Recorded sound was introduced over a century ago but this only allowed for mono recordings.

 This paved the way for the future development of sound recording and led to some rapid developments in the science of sound recording.

One key innovation was the creation of the thermionic valve by Lee De Forest (1907). The invention of the valve enabled the amplification of electrical signals, offering the possibility of recording multiple signals, combining them and controlling them.

 After this development, there was an increased amount of research on how best to record and reproduce a more effective sound field.

 Stereo was the next significant breakthrough with innovations in techniques implemented by Alan Blumlein where he used stereo coincident techniques in an attempt to accurately represent a more realistic sound stage in broadcasting.

The first major surround sound experience emerged in the 1940s with Disney’s Fantasia movie.One of the most significant developments in the sound creation for this movie was the development of the pan pot which would be used to place the individual discrete channels to a specific area within the soundfield.

 Modern techniques were also implemented in the making of this film such as spot microphone placement, overdubbing and click tracks. Surround sound technology didn’t progress much further after this and it wasn’t until the 1970s that quadraphonic sound emerged and was aimed at the consumer market.

Problem With Quadraphonic Sound

Quad sound employed 4 speakers which comprised of two conventional speakers at the front and two at the rear. The speakers would be at an angle of 90 degrees to each other.

 This set up created major problems in trying to create a stable surround image. It was found that creating this system with little or no scientific research on the psychoacoustics of the system had led to problems in creating a stable stereo image as well as the side information which wasn’t being replicated effectively.

 The listener would have to stay in a small sweet spot area to gain any real benefits of a surround experience.

 Quad sound was introduced to the consumer market in the formats  of reel to reel and 8 track cartridge, catering for 4 channel systems. The majority of quadraphonic systems used signal matrixing to create the desired effect.

 Matrixing would attempt to combine the 4 quad channels into two channels. This would help the quadraphonic two channel recording to be compatible with domestic stereo systems, allowing consumers to play their quad, stereo and mono records from the same system.

 There were various flaws to this system however. The main problem was that it was impossible to recreate the four separate channels effectively. There was no way to isolate the individual channels and artifacts could be introduced into the overall signals such as crosstalk between channels, making accurate stereo imaging difficult.

 With these technical problems as well as the fact that there were numerous incompatible sound systems, confused the consumer and led to the demise of the quadraphonic format.

Current State of Ambisonics

There have been several further developments in Ambisonics over this period of time as digital technology emerged, introducing formats such as CD DVD video and systems such as 5.1 surround.

 One of the most important developments in Ambisonics emerged when it was reasoned how best to develop a system that could be made compatible with the emerging 5.1 surround systems. Dr Geoff Barton was the co founder of a TV production company involved in satellite TV services.

 Barton’s proposal, was that the 5.1 system could be utilized by taking the loud speaker feeds from an Ambisonic decoder in the actual studio which could transfer to the listener within their home environment.

The system wouldn’t be strictly Ambisonic as some compromises would be made to make both systems compatible. One of the major compromises, was to ensure that the listener at home had their speakers in exactly the same positons as the speakers in the studio environment which is a situation that exists in 5.1 systems but not in an Ambisonic system which allows the listener to place their speakers as they wish without compromising the authenticity and integrity of the original studio signal.

 The decoders used in the studio and by the listener at home would also differ but, overall, it appeared that Dr Barton had created a workable solution. G format , is the term now recognized to describe a system solution and was outlined in a paper report produced for the AES.

 There are also products released on the market which allow sounds to be recorded with an Ambisonics microphone array and then to subsequently decode those recordings and render them suitable to be transferred into a 5.1 surround system.

Soundfield are one company that has been prominent in this field and produced the SP451 sound processor.

 This rackmount processor and dedicated decoder , takes the B format signals from a Soundfield microphone and produces the D format signals which would then transfer to the 5.1 surround system now recognized as the G format. This system is advantageous for the listener as a decoder is not required at the listeners end although the height information in the 5.1 system will not be replicated and the speaker positions for the listener still have to conform to the conventional surround sound mix.

The future of Ambisonic sound is filled with potential. Multi media technology such as home cinema and gaming technology has become open to the mainstream consumer market. With immerssive interactive experiences now being demanded by consumers and 3d visuals being a possibility in the home.The potential for Ambisonic sound to be used in these areas is huge.

Could it be a matter of time before a forgotten technology has a re birth and is a staple ingredient of how sound can enhance multi media entertainment?

 This post is the first of many that will concentrate on loads of different areas of sound design and music production.From recording bands to creating sound for film and computer games, i'll try to cover a lot of aspects of music production hope fully some of it might even be interesting.

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