Saturday, 4 May 2019

Fairlight CMI IIx - Page 8 - Sound Sampling

The Fairlight synthesizer is possibly most famous for being one of the worlds very first digital samplers.
Those early sounds from 1979 have even been immortalized by being added to the NFSA's "Sounds of Australia". (National Film & Sound Archive)

We take sampling for granted these days , but in the 1970's and 80's  it was a revolution.

Page 8 is where the sampling magic happens.

Here is the HELP page for page8


To access page 8 type
P8<return>

To access page 8 help type
H8<return>
To step through the help pages use the ADD & SUB buttons on the keyboard.

 Here is a typical view of Page8.

Sampling on the Fairlight results in the creation of a CMI voice waveform via ADC (analog to Digital conversion).
The sounds are sampled from the rear of the mainframe.
There you will find 3 sockets: Mic in, Line in and ADC direct. and some input selection switches.

A new blank voice must be created 1st. It must be in Mode 4.
NPHONY must be at least 1.
Type :       C,r,filename<return>

                  where r = register A - H
                 filename  = 1-8 character voice filename
                 Example: C,A,JONO2<return> creates a file named JONO2.VC in register A
                

See Pages 2 & 3.

Next load this voice.
Eg: L,A,JONO2<return>

Typing "S <return>" or touching "Sample" will start the sampling process .
After about a second, the display box will show the sampled waveform.




The Filters
The 2 filters are digitally controlled. They filter out unwanted high or low frequency components
of the external sound to be sampled.
They range between 1 to 9.
Their cutoff points are set by Filter Low & Filter High.
The filter circuit uses a CMOS 4051. It takes the form of a switched resistor (SR) filter.
SR filters employ resistors with analog switches to control frequency cutoff.
You will find this circuit on the Master Card.
There is an output at the back of the CMI (filter out) to monitor  the effect of any filtering.
These filters are by-passed when using the ADC direct input.

The Sample Level is a software based volume control. It can be used to attenuate input signals if they are too "loud".
Range : 0 to 255.
Set this to a level where the loudest section of the sound is neither clipping or less than 80% of full scale.

The Sample Rate:
This is the frequency at which the external signal is sampled and then converted. thus determining two aspects of the sample: Bandwidth & Duration.
BANDWIDTH is also called Fidelity. The higher the sample rate, the better the bandwidth.
DURATION of the sample gets shorter with increasing sample rate.
Thus you can see there is a trade off between the two. Higher bandwidth will result in a shorter duration.
eg: a sample rate of 16384Hz  = a sample with 1 second duration
      a sample rate of   8192Hz  = a sample with 2 seconds duration.


 Some natural sounds contain complex phase relationships varying in pitch & volume. This can make them difficult to record. Thus recording live can be difficult. The manual suggests recording these sounds to tape first.

The Trigger (Level & Delay)

The Level ranges between -128 to + 127
This is the amplitude threshold used to set the point at which sampling starts.
When you hit "S<return>", the CMI will wait for this level to be reached before it will start recording.
A trigger level of 0 (zero) will result in sampling commencing immediately after S<return> is typed.
This is called manual triggering.

The Delay
If desired, you can use this to set a delay after the trigger has occurred.
The range is 0 to 65535 msecs (0 to 65.5 secs).

After sampling, use page 6 to inspect for clipping.
You can also do things like rotating, zeroing, mixing etc etc.

To interrupt sampling at any time use:
<Ctrl><esc> (hold down ctrl and press esc)

To display the waveform type:
D<return>

You can now use Page D to view your sample:


''''''''''''''''''''''''''
About the ADC
The actual analog to digital conversion is done by a 10 bit converter, even though the CMI is an 8 bit machine.
Only the top 8 bits of the sample values are stored.
The two LSBs (least significants bits are ignored).

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Friday, 3 May 2019

Fairlight CMI IIx - Page 6 - Waveform Drawing

Page 6 is all about displaying, manipulating and drawing waveforms.
You can draw your own with a light pen or use a sampled or saved waveform.
Then you can squeeze, invert, rotate, reverse, merge, mix , etc etc waveforms.

Here is the HELP page for page 6


To access page 6 type
P6<return>

To access page 6 help type
H6<return>
To step through the help pages use the ADD & SUB buttons on the keyboard

Here is a typical view of Page 6.
Looks a lot like an oscilloscope

It's a view of the 1st segment of the voice/waveform AAATEST.VC

Below is segment 8

Below are more segments ... 14 & 66

The CMI divides the waveform into 128 segments.
To set the current segment number use  <arrow> keys to tab to the needed segment


Here is the same waveform in Page D
The segments are stacked one in front of the other

Pages D, 4, 5 & 6 are connected ... the 1 waveform will be represented differently in each page and changes made to the waveform in one page will effect the other pages.

The Waveform Graph

The waveform graph displays 1 segment.
Each segment is made up of 128 points

Waveforms can be drawn or altered using either the light pen or the alpha numeric keyboard.
The latter method uses the concept of points and levels.
We assign LEVELS to individual waveform points. The horizontal divisions mark 1, 8, and 16 POINT intervals across the graph.

Points range from 0 - 128
Levels range between -128 & +127

Use the <arrow> key or light pen to tab to POINT / LEVEL.

Theoretically, you could type in the level for each point of each segment........ and repeat this for the entire waveform. This would be a huge and tiresome task though it is very precise..
The light-pen is a much easier option I think.

Join Vs Plot Switch

Join - J<return>         Each point is joined to the last by a straight line.
Plot - P<return>        Each point can be altered without effecting the others

Note.
The waveform is not finalised until the FILL command is entered.


Notice the two faders.
These control the stepping rate & the segment number to be displayed.

To display a segment type:
D<return>            displays the current segment
D,s<return>         displays segment (s)
D,s,e<return>      displays segment (s) through to (e)...... s= start, e= end (range 1 to 128)
D,*<return>         displays all segments

When a range of segments is specified as in "D,s,e<return> "
they will be displayed sequentially... called stepping display.

To start/stop the stepping display type:
S<return>           to start
S<return>           to stop when in progress.
The fader sets the time for which each segment is displayed.


The segment section covers:
Display, Zero, Fill and Invert.

Invert simply inverts the phase of the waveform.
For example, the command :    I,85,95<return>
This will INVERT (and automatically FILL) segments 85 to 95.
The ZERO command will zero the waveform and create a NULL voice
For example to silence the waveform:

Type:
Z<return>
F,*<return> ...... this second command will fill all segments.

Page6 can generate 3 traditional waveforms with the click of a button (or light pen).
Triangle     TRI<return>
Saw            SAW<return>
Square.       SQ<return>

Pulse width is set using:
n<set>

Note that the waveform is not affected until you hit FILL
To generate a sine wave you need to use page 5. Use the harmonic faders.


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Buchla Index

Buchla, Non-Buchla DIY Clones, Buchla Format & Ecosystem Modules


General


Easel
 + Buchla Easel - Program card map for BEMI,Vintage & Rev II 208. 
 + Korg Monotron (duo) -Buchlaized 
 + Portabellabz BOB Expander (Papz/Constantin) - build notes etc
 + Portabellabz BOB Toolbox - (Papz/Constantin) - Build notes 
 + Portabellabz BOB -VCS3 card - Every Easel needs a synthi -build notes
 + DIY Audio mixer for 208 easel - Schematic 
 + Kaitlyn Aurelia Smith - SugarMountain 
 + Variation 3 
 + Volca Bass
 + Buchlaized Korg Volca Bass 
 + Korg Monotron 
 + Easel Program Card Map 
 + Krell patch - figure 21

100 
 + 102 - Non-Buchla DIY - LA 67 - build notes
 + Phase vs Frequency Shifting - Model 196 vs 185


200
 + 25S
      Non-Buchla DIY - 216r - build notes
 + 246r - sequencer (Non-Buchla DIY)
 + 248r
      + 248r expander (rev1) 
+ 281
 + 284r - Build pics 
 + 296r - Roman/Buchla format sound demo


200e
+ 222e
      *Notes
      *Buchla Grunge
      *Buchla Acid Train
+ 225e
+ 250e
+251e
       *  My notes on the 251e
+ 252e
      * Notes on the 252e
      * Buchla 252e & 266r
+ 261e
+ 266e
       * Notes on the 266e 

Sunday, 28 April 2019

Fairlight CMI IIx - Page 5 - Harmonic Amplitude Profiles

Page 5 of the Fairlight is (like page 4) about Additive Synthesis.
A Fourier analysis command allows sounds to be converted to harmonics

The above pic shows a loaded voice file. (BALDY1.VC)
This typical page 5 display shows the harmonic overtones as a set of 32 faders.
It's a bit like a graphic equaliser.
The faders are logarithmic (range is 0-255)
First load the voice by typing L,A,name<enter>

To access page 5 type
P5<return>

To access page 5 help type
H5<return>
To step through the help pages use the ADD & SUB buttons on the keyboard


Though both page 4 & 5 deal with additive synthesis and harmonics, they approach it in different ways.
Page 4 can change one harmonic over all waveform segments at once.
Page 5  can change all harmonics over one waveform segment at once.


After loading the voice you can move the faders

Notice how the waveform shape has changed from the original voice file at the very top pic.
This is after typing the command: COMPUTE

My lightpen is misbehaving but I can still get around the page using the arrow keys'
More help screen info:

If you wish to use the keyboard to set the fader type:
n<set>
where n = 0 to 255
The faders are log in function.
The top of the fader = 255
Half way up the fader = 16

You can also use the ADD & SUB keys.
To zero all faders type
Z<return>

---------------------
Setting the mode of a voice in Page 5.
This must be done in Page 3

This is a pic of page 3
Usethe arrow key to tab to MODE for the desired voice.
Type: n<set>
n = 1 or 4

Read page 4 for more info about mode 1 vs 4

tHE mode chosen in important.
A Mode 1  sound has 32 sets of faders
A Mode 4 sound will have 128.

The current segment number will be indicated on the display with a small pic of the "wave slice"
in the lower right corner.
Here is what segment 1 looks like.
Use the arrow keys to tab to "current segment" on the screen.
Type n<set>
where n = the segment number you wish to see.

You can quickly jump through segments by using the ADD and SUB keys.

Here is segment 6

Here is segment 32

Each segment is unique.
The synthesized sound changes throughout its duration.

You can compute a different waveform for each segment.
The COMPUTE command mathematically processes these to generate a waveform by additive synthesis.
The page 4 profile graph and page 5 faders are linked.
If you change one, this change will be reflected in the other.

Here is the fader page  of the voice AAATEST.VC


Here is its corresponding page 4



This is what page D looks like for AAATEST.VC


To be continued..............

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Saturday, 27 April 2019

Trautonium - The birds -Peter Pichler

Thanks Peter for a wonderful performance.
https://www.peterpichler-trautonium.com/english/



This is the first time a Trautonium has left Europe.
Peter took these instruments to Perth, Canberra and Melbourne.
Sadly not Sydney. So I made a special trip to Canberra to see this.










Friday, 26 April 2019

Fairlight CMI - Page 7 - Control Parameters

This is a really important page.
Here you can modify and control various aspects of a sound.
Things like volume, looping, attack, sustain, portomento, the mode (page 4), etc..

To access page 7 type
P7<return>

To access page 7 help type
H7<return>
To step through the help pages use the ADD & SUB buttons on the keyboard  


A typical page 7 looks like this.

My lightpen is misbehaving at the moment (I'm hoping its just the graphics card), so I'm using the alpha-numeric keyboard to get around.
You use the arrow keys to move around the screen


type the numerical value you wish to enter, then hit the SET key on the lower right.

The music keyboard has 2 switches and 3 sliders (called controls):

These switches & sliders control some the parameters on the lower right of the screen.
There are in effect 6 sliders (called controls)  &  5 switches.
The other sliders & switches are controlled with footpedals that can be plugged into the rear of the keyboard.


Here is a page 2 example of a disk containing lots of voice files.

Notice that in addition to the .VC files we also have .CO files with the same names.
Eg we have a AH3.VC file along with a AH3.CO file
The .CO file is a command file. This has been linked to the voice file (using the command LINK).
So when this voice is loaded into the CMI, it will pull in its linked & specified control file.
Control files may also have different names from their intended voices.



The filter is I understand to be LP. There is one on each channel card. The cutoff frequency is raised simply by increasing the value.

Mode: see page 4 & 5.

KeyLevel: This controls the key velocity of the music keyboard. This can be patches to Level and Attack

Portamento is as you'd expect ... produces glides between each new pitch. The rate of note glide is set by the speed control.

Glissando differs from Portamento in that the glide is not continuous. Rather it's chromatic. All the notes between the start and end notes are played. If both are selected, portamento takes precedence.

Constant Time: 
This is a switch that selects between two types of glide (constant glide vs changing glide).
Constant glide /time  (on) = the same time is taken to travel any musical interval. This results in polyphonic portamento or glissando.
Constant glide /time  (off) = the rate of change is determined by SPEED. The time taken to glide varies with the size of the interval.
Attack & Damping

Attack has a range of zero to 16,384 milisecs. You can patch this to the KEYVEL to control attack time.
This works only in mode 4.

Damping has a range of zero to 65,536 milisecs
This is reduced to 16,384 in mode 4
Damping = decay time of the voice.
If loop is activated, the voice will be looped until the damping time expires

Loop
This is where we can set loop points.
Any voices are cut into 128 segments.
LOOP CNTRL = on, off (uses a switch - set it from your keyboard or pedals).
                            This turns the looping of your sound on & off.
LOOP START =  range of 1-128.
LOOP LNGTH = length . range of 0 - 128.
The actual loop points are displayed graphically on page 4.
Page 4 --- Harmonic Profiles 





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Thursday, 25 April 2019

Fairlight CMI IIx - Page 4 - Harmonic Profiles

Page 4 of the Fairlight is all about Additive Synthesis.
To access page 4 type
P4<return>

To access page 4 help type
H4<return>
To step through the help pages use the ADD & SUB buttons on the keyboard 


The manual introduces this page as:
 "offering facilities for rapid additive sine-wave Fourier synthesis by
harmonic amplitude manipulation."

Here is an example of a page 4 display


 Page 4 (and page 5) permit the drawing of 32 sine wave harmonic amplitude envelopes, duration envelopes, and master volume envelopes.
The harmonic envelope is similar in concept to a ADSR envelope you will find on many synthesizers.

The loop feature means that any part of the overall waveform can be made to loop (sustain).
After you draw your waveforms, the CMI will process the data and yield the final waveform by the COMPUTE command. (use either the lightpen or the alpha numeric keyboard).

Here is the first HELP sheet for page 4


There are many forms of synthesis.
Most of us are familar with subtractive synthesis where you start off with a waveform that is rich in harmonics such as a square-wave and you subtract harmonics using devices such as filters.

Additive synthesis does the reverse. You start off with the simplest building block of sound , the sine wave & you add more harmonics. In theory, you can construct any waveform using just simple sine waves.

The mathematician Joseph Fourier first proposed this theory back in the 19th century.
The concepts of additive synthesis had also existed with the invention of the organ. This instrument uses different pipes of varying pitch. These can be combined to create very complex sounds & timbres.

The harmonic series works like this:
If harmonic 1 = fundamental pitch.(FP)
Harmonic 2 = FP x 2 = harmonic 1 x 2
Harmonic 3 = FP x 3. etc etc

You need to add lots of sine-waves to produce waveforms that closely approximates idealised mathematical forms. So maybe this method isn't the best for this.
The strength of additive synthesis lies in the fact that you can exert control over every partial component of our sound. This can produce some very intricate and wonderful results.

MODES

Page 4 has 2 modes:
Mode 1 & Mode 4.
The type of sounds you want to make will help you choose which mode to use.


In Mode 1 each vertical division represents 1 waveform segment
This means there are 32 waveform segments.
Each segment in turn is sustained according to the DURATION profile position of the segment.
During this time, the amplitude is modulated according to the slope of the ENERGY profle.
It's good for doing harmonic synthesis & generating "electronic" sounds. (pages 4 & 5)

In Mode 4 each vertical division represents 4 waveform segments.
(I guess this is why its called mode 4)
This means there are 128 waveform segments.
Each segment is played once before moving to the next segment.
This is more aimed at drawing & sampling. The manual suggests it be used when making "natural sounds" .
(pages 6 & 8)

 The numbers 1 to 32 are actually lightpen switches to display all 32 waveforms. However, if this is done the display can look confusing, so its possible to hide the display of some of the waveforms (they will still remain active, and will contribute to the final sound).

To set the mode of a voice you need to use page 3.
Here is a typical page 3

Use the arrow key to tab to MODE for the desired voice.
Type: n<set>
n = 1 or 4

---------------------------------------

Wave Table Voice Facility

This option allows you to use waveforms other sine waves for additive synthesis.
Waveform segments from another loaded voice can be the source of the sound.

Harmonic numbers, now become segment numbers. You can't however select the current voice as a wave table.

In order to use this voice facility you need to load a voice into the system's memory first.
This is done in page 3.  After you have done this, you will need to select in within page 4.

To select any loaded voice for display & modification (within page 4) type:

V,filename<return>      or               V,n<return>

where filename  = the name of the loaded voice.
                      n  = the number 1-8 of a loaded voice.

The name, number, and mode of the current voice are shown in the upper right corner of the page.
The names and numbers of all loaded voices are shown on page 3 & page 7.

 
--------------------------------
This page is constantly being updated as I explore the Fairlight system.
If you have a Fairlight, don't hesitate to let me know if there are any mistakes or omissions.

 To be continued...................

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