Month: December 2023

Ping-Pong (Cross) Echo.

25/12/2023 / Stephen / 0 Comments

A Ping-Pong Echo is much like a normal echo, except that its a stereo effect and the left signal feeds back to the right channel, and the right channel feeds back to the left channel, bouncing the signal back and forth across the stereo field.
Hence the term ping-pong delay. Shown below is a schematic diagram of a cross delay.

Basic block diagram of a ping-pong delay

Creating our Ping-Pong echo in SynthEdit.

To create a Ping-Pong Echo in SynthEdit we must use an external feedback path rather than the Delay2 module’s internal feedback circuit and as you know if you try to create this structure with a normal feedback path, you will get the error message: “This patch contains a FEEDBACK path, Please remove.”
We get round this problem using a pair of Feedback – Volts modules to create feedback paths that are acceptable to SynthEdit.
See the layout below:

For the sake of simplicity I have kept to one set of controls for both left and right channels, but there’s nothing stopping you from having separate controls for left and right channels.
Of course you can build on this to add HF Damping into the feedback loop, displays to show the delay time etc.

Delay in SynthEdit.

21/12/2023 / Stephen / 0 Comments

The Delay2 module is a handy module which introduces a delay into an audio signal. The Delay2 Module has a minimum of 0 seconds, and a maximum of 10 seconds delay time.
To control the delay time of the delay module, you use it’s Modulation input to adjust the delay time parameter set in the modules property view. The Delay Time parameter sets the maximum delay time of the module. This maximum time can then be modulated, or adjusted, from 0 to the maximum of 10 using the modulation input. This allows you to make an adjustable delay by connecting a slider, or by modulating the input using another control voltage source such as an oscillator module, we can create effects such as flanging and chorus.

Very basic delay effect.

To set up a delay time that you can vary vary from 0 to 1 second:
1) Connect a slider module to the Delay module’s Modulation plug.
2) Go to the delay module’s properties panel and make sure the Delay Time value is at the default value of 1.0 Second.
The feedback plug gives you control over the internally generated feedback level, between 0% (0 volts) and 100% (10 volts).
Note: The interpolate out option is only used when you are going to have rapid changes in the value on the Modulation plug to prevent unwanted clicks, and stepping effects. If you’re not modulating the plug with an Envelope or LFO leave it un-ticked to save on CPU power.

This works, and it’s relatively easy to work out the delay time from the slider’s readout; 10V gives us a 1 second delay time , 5 Volts give us a 0.5 sec delay, 0 V gives 0 Sec, but its not the best of solutions.

Adding a Delay time display in mS.

Normally, delay times are given in milliseconds, so how do we get our slider to display the actual delay time in milliseconds? Using the structure below it’s quite easy;
1) Set the maximum value of the delay slider to 1000, leaving the minimum at 0,
2) Connect a Multiply module between the slider and the Delay2 Modulation plug. Set input 2 to 0.01 either in the Properties panel or using a Fixed Volts module to supply the value.
3) For the display we use a Volts to Float2 converter, feeding the float value to a
PatchMemory Float Out3 where it’s converted from DSP to GUI data.
4) The GUI float value is then converted to Text using a Float To Text module, mine is set to 0 decimal places, but it all depends on your delay time range, and how accurate you prefer to be.
5) Untick the Writeable property for the Text Entry4 module, as changing this value won’t affect the delay time (this would only frustrate your plug-in’s users).
We now have a readout in milliseconds. Well yes, you could use the readout on the slider control but this has a small drawback. By default it adds three decimal places, and with the sort of values we’re using the display is cut off at one end.

Adding a readout for the delay time in milliseconds

How our mS display looks:

Adding damped feedback to an echo effect in SynthEdit.

Again, the project above is a good basic Echo, but we would often find that in real life the sounds that are “echoed” will gradually change, they will lose some of their high frequency elements. What can we do about this?
Well, with the structure below we can add an external feedback loop, that has a 1 pole low pass filter included to give us variable HF damping. Because we are using a Feedback (Volts) module we don’t get error messages about feedback loops not being allowed, and the slight delay introduced by the module will have no real effect on the feedback. In practice we won’t want the filter to go as low as 0 hZ, so we could set the Filter to kHz per volt and the slider range from say .2 to 8000, giving a range of 200Hz to 8Khz. You’ll need to set some decimal places for the Float To Text conversion module to get a readout of both Hz and kHz.
Note: When testing in Synthedit due to the displays being GUI modules you’ll only see the Delay time and Damping frequency displays update if the Audio Engine is running.

How it looks with the filter control and frequency readout.

Bitcrusher (reducer) in Synthedit.

15/12/2023 / Stephen / 0 Comments

About Bitcrusher effects.

A Bitcrusher is a low-fidelity effect used to emulate the distortion introduced by older samplers, and wavetable synthesizers. They work by reducing the sampling rate (the frequency at which the audio signal is sampled, or bit-rate), or reducing the bit size of the samples (the number of bits used to digitally represent the amplitude of the audio waveform). These can reductions can be used independently or together. A Bitcrusher will by virtue of the way it operates introduce digital noise and aliasing into the audio, this is deliberate, as this is emulating the sound of the older low resolution audio samplers and wavetable synthesizers.

Bitcrusher options in SynthEdit:
There are currently three options;
1) A third party module by Elena Novaretti in her ED DSP module pack, the
ED Crusher which (personal opinion) I think is currently the better option. This has the edge on the community modules BitCrushers in that it allows reduction of both bit depth, and sampling rate together.
2) The BitReducer in the Community modules.
3) The BitCrusher in the Community modules.

A basic SynthEdit Bitcrusher design.

The 1 pole LP in the input is fixed at 15kHz, and is included to limit the input signal to prevent too high a frequency entering and causing un necessary aliasing or distortion. The sampling frequency (X Quantize) is controlled via the Volts to Float2 module, and the properties for this module are set to Volts DC (Fast), and an update rate of 10 Hz.
The amplitude sampling (Y Quantize) setting is controlled via a RoySwitchL(Int), and a set of Integer fixed values (4, 8, 16, 32) selectable via a drop down list.
The LP filter in the output is not an attempt to reduce aliasing (once you introduce aliasing you can’t get rid of it) as it’s a part of what we are re-creating, it’s there to reduce the high frequency output to get closer to the low fi sound of old samplers and wavetable synthesizers.
Note: All Bitcrushers produce noise and aliasing because you are reducing the sampling rates, and thus decreasing the Nyquist frequency.
Note: You can put chords through a bitcrusher, but it sounds much better with single notes. Once you start putting chords, or harmonically complex signals through this effect the resulting audio quickly becomes a noisy mess (very much like using an ordinary fuzz box).
If you’re using this effect in a synthesizer plugin to recreate the sound of a vintage 8 bit wavetable synthesizer it’s much better to have one bitcrusher per voice, than one bitcrusher used on all the voices together.
Will oversampling improve the sound from a bitcrusher? In short, no. You will actually be wasting CPU resources on something that will not help create the lo-fi sound that’s intended. You won’t reduce the noise or aliasing noticeably…and I have to ask why you want to increase the quality when the aim is to create a vintage lo-fi sound?

Quantization in SynthEdit

12/12/2023 / Stephen / 0 Comments

Quantization is used to constrain a control voltage to discrete steps. The module takes the amplitude of the input voltage and breaks it down into the steps specified by the voltage on the Step Size control plug. Quantization is used on control voltages for pitch an filter cut-off voltages. (No reason why you can’t apply it to audio, but the correct module for audio is a Bitcrusher which gives you better control). Quantization is often used for generative and ambient music.
The Synthedit Quantizer will work on Audio signals, and progressively reduce a sine wave signal form the original sine wave, through a stepped sine wave to a pulse waveform, which is quite different in effect from a bit-crusher (bit reducer).
Note: all a quantizer will do when applied to audio is to introduce distortion, it does not introduce any frequency changing or shifting effects.

Plugs.
Left Hand Side:

Signal In:- (Voltage) Control Voltage Input signal
Step Size:- (Voltage) Quantisation step size in Volts

Right Hand Side:

Signal Out:- (Voltage) Quantized Control Voltage output.

The screenshot below shows the effect of quantizing a +/- 5 Volts sine wave. Note: In some cases the peak-to-peak voltage of the output will actually be greater than the =/- 5 Volts input to accommodate the correct voltages between each step.

An improved Reverb design in Synthedit .

12/12/2023 / Stephen / 1 Comment

Using a single Delay2 for a very basic reverb effect is all well and good, but has its limitations, one of which is that at the short delay times we are using there can be noticeable peaks and troughs in the frequency response, as what we are setting up is basically a comb filter, so we can get some ringing effects on transients at certain frequencies, and also a “drainpipe” like effect on other sounds. Not good. So what’s the answer?

The Schroeder Model of Reverb.

If we put a sharp transient sound such as a handclap into a delay line then
exponentially decaying impulses follow the first impulse. Though this
is similar to an exponentially decaying reverb, in the frequency
response peaks occur at equally spaced frequencies like the teeth of a
comb, hence the name comb filter.

How sound decays (fades away) in a room or hall.

This will result in a ringing metallic sound, especially once feedback is added into the mix. Digital audio pioneer Manfred Schroeder proposed using parallel delays with differing delay times, in combination with all-pass filters for digital reverb. His idea proposed a structure with four parallel delays and two serial all-pass filters. The delays create the reflections, and the all-pass filters “smear” any transients thus making the reverb more diffuse and reducing any resonances. Schroeder’s design is shown below in block form. Each comb is a Delay2 module.

Block diagram of the Schroeder type reverb

The complete Schroeder emulation in SynthEdit

6 x Delay container:
A set of six Delay2 modules pre-set to differing delay times within the range of 10 to 60 mS which is the ideal range for reverb, any longer and we start to get an echo effect, and shorter and there’s no noticeable reverb effect.
All the outputs from the Delay2 modules are fed into the Input 1 of a Divide module, this is because if we just feed all the signals into a normal module they will all be added together, more than likely exceeding the input range of the module causing clipping. Input 2 of the divide module is set to 6 (the number of delay modules) to restore the normal signal range. There have been various calculations made for the “ideal” delay times, but in practice a natural reverb depends on the shape and size of the area, and the things in that area.
There has been a calculation made of the “ideal” delay times which are
50, 53, 61, 68, 72, and 78 ms

The structure of the 6 X Delay container

All Pass Container:
The All Pass container uses two more Delay two modules in series one set to 17mS, and another to 74mS these two delays give an added “blurring” effect to any transients, along with a 1pole Low Pass filter for variable HF damping on the reverb “tail”

Feedback Container:
The feedback container has a Feedback – Volts module to allow feedback. A combination of the inherent time delay in a feedback loop in SynthEdit, and the two All Pass filter modules adds further frequency dependant phase shifting to further reduce ringing and metallic sounding transients. Again the frequencies of the All Pass filters are fairly random, but spaced out to try and give maximum effectiveness in reducing unwanted audio artefacts.

Creating a very basic Reverb effect in Synthedit with the Delay2 Module.

10/12/2023 / Stephen / 0 Comments

BBD Reverbs.
First there were plate and spring line reverbs, then along came an electronic “chip” called a Bucket Brigade Delay (BBD). This used high impedance CMOS devices and capacitors to pass the electric charge (audio) along the line, each stage in the line adding a small delay. These worked well, and eliminated the problems of external noise entering the signal chain. However the way they worked introduced their own artefacts such as aliasing at high frequencies, filtering was used filtering to restrict this, and they also introduced a certain amount of distortion, which gave them their own unique low-fi sound. By using multiple chips with different delay times you could get some quite complex reverbs.

Delay2.

Creates a delay (echo or reverb) effect on audio signals very similar to BBD chips.
Note: If you intend to modulate the delay time, you should enable interpolation, this reduces clicks and “zipper” noise.

Plugs.
Left Hand Side:
Signal In:- (Voltage) Input Signal
Modulation:- (Voltage) Varies the delay time dynamically (0 to 10V)
Feedback:- (Voltage) Controls the amount of feedback of the delayed signal.

Right Hand Side:
Signal Out: – (Voltage) Audio output signal
Parameters:
Delay Time (secs): – Maximum delay time in Seconds. The delay time is limited to a maximum of 10 seconds.
Interpolate Output: – Provides smoother modulation of delay time, but with an increase in CPU load.

Basis BBD style, single delay reverb effect.
We can emulate this in a SynthEdit module, which takes the audio input, and much like the old BBD chips, slices up the audio into samples and passes them down the delay line.
Shown below is our very basic reverb effect.
Delay Time:- Set the delay time to 0.001 Seconds (10mS) in properties
Delay control:– Set the maximum voltage for the delay slider to 0.06 volts, this gives us a delay time range of 10mS to 70mS.
Feedback:– Set the maximum voltage for the Feedback slider to 8 volts, as we don’t want the absolute maximum feedback level as it will just give uncontrolled oscillation.
Wet/Dry:- This control will default to a range of -5 volts to +5 volts. Leave this as it is to give the full range of wet to dry effect.

There’s one important point to note, if this reverb is being included as part of a Synthesizer VST we need to think about polyphony. It really isn’t needed for a reverb effect so we need to containerise our reverb, put a Voice Combiner module in place in the input- the voice combiner forces then the Modules in this container to stay Monophonic. Otherwise every time you play more than one note on your Synth plug in it will create an un-necessary clone of the reverb module wasting CPU and memory.

Making the Reverb Stereo.

Converting to Stereo is easy, we just create two identical Reverb chains, one for the left channel, and one for the right channel. Keep the two delay and feedback levels linked otherwise some very peculiar (and unwanted) phase and stereo imaging effects can be introduced.

Limitations of basic Delay reverbs.

This is structure is OK for a very basic reverb effect, but has its limitations, one of which is that at the short delay times we are using there can be noticeable peaks and troughs in the frequency response, as what we are setting up is basically a comb filter, so we can get some ringing effects on transients at certain frequencies, and also a “drainpipe” like effect on other sounds. Not so good. So what’s the answer?
Fortunately there is one that was devised by Manfred Schroeder, known as the Schroeder Model of reverb.

Reverb. What exactly is it, and how can we create it in SynthEdit?-The basics.

08/12/2023 / Stephen / 0 Comments

Reverberation or reverb, in music is a persistence of sound, or a very short echo after a sound is produced. Reverb is created when a sound or signal is made inside an enclosed or semi enclosed area, the reflected sounds from the enclosing walls causing complex sound reflections to build up and then decay as the sound is absorbed by the objects in the space – which could include furniture, people, walls etc. This is most noticeable when the sound source stops and the reverberations continue, their volume decreasing until zero is reached. Reverb is pitch dependent, as sounds of different pitches will arrive back to the listener at slightly different times, and unless the room is very carefully designed it will have a resonant frequency at which sounds will be emphasized.
In comparison to a distinct echo, which usually 50 to 100 ms after the previous sound, reverb is made up of reflections of sounds that arrive in less than about 50 ms. As time passes, the level of the reflections gradually reduces to non-noticeable levels. Anywhere there are surfaces to reflect sounds from you get reverb. The more complex the shape of the room and the more objects in it the more complex the reverb is. Certain frequencies can be boosted, and others cut due to the wavelength of the sounds interacting with the size of the room (resonance). It may be created through physical means, such as echo chambers, or electronically through audio signal processing. There are various means of achieving a reverb effect listed below.

Echo chambers
The first reverb effects, introduced in the 1930s, were created by playing recordings through loudspeakers in medium to large spaces (various spaces such as empty rooms, bathrooms and even stairwells have been used) and mixing the sound with the original using strategically placed microphones.
The American producer Bill Putnam is credited for the first artistic use of artificial reverb in music, on the 1947 song “Peg o’ My Heart” by the Harmonicats. Putnam placed a microphone and loudspeaker in the studio bathroom to create an echo chamber, adding an “eerie dimension”.
The first two examples of reverb (Spring and Plate) are included just to give some historical background as we are going to be using digital sound processing to achieve our reverb effects.

Plate reverb
A plate reverb system uses an electromechanical transducer, similar to the driver in a loudspeaker, to create vibrations in a large plate of sheet metal. The plate’s motion is picked up by one or more contact microphones. The audio signal from these is then mixed with the original “dry” signal. Plate reverb was introduced in the late 1950s by Elektromesstechnik with their EMT 140 design. The greatest problem with plate reverb units is their size and weight, which limits their use to recording studios.

A spring reverb.
Spring reverbs, were introduced by the company Bell Labs, using a set of springs mounted inside a box. They work in a similar way to plate reverb, with a transducer at one end of the spring, and a pickup placed at the far end of the spring, to reproduce more realistic reverb different length springs could be mixed together (longer spring = longer delay time) .
They can have a very distinctive “twangy” sound with loud percussive sounds due to the springs. One major drawback is sensitivity to outside vibration.
They were popular in the 1960s, and were first used by the Hammond company to add reverb to Hammond organs.
They became popular with guitarists, including surf musicians such as the Beach Boys, and spring reverb could easily be built into guitar amplifiers.
They were also used by dub reggae musicians such as King Tubby. The American engineer Laurens Hammond of the Hammond company was granted a patent on a spring reverb system in 1939.

Digital reverb
Digital reverb units simulate reverb by using multiple delay lines that have different delay times and variable feedback, giving the impression of sound bouncing off of multiple surfaces. Some digital effects allow users to independently adjust early and late reflections. Digital reverb was introduced in 1976 by EMT with the EMT 250, and became increasingly popular with many groups and studios in the 1980s.

Gated reverb
Gated reverb combines reverb with a noise gate, creating a “large” reverb sound with a short tail (the tail is cut short by the noise gate).
It was pioneered by the English recording engineer Hugh Padgham and the drummer Phil Collins, and became a staple of 1980s pop music.

Convolution reverb.
Convolution uses impulse responses to record the reverberation of physical spaces and recreate them digitally. The first real-time convolution reverb processor, the DRE S777, was announced by Sony in 1999.
Convolution reverb is often used in film production, with sound engineers recording impulse responses of sets and locations so sounds can be added in post-production with realistic sounding reverberation.
Basically, a convolution reverb takes an input signal (the sound to be reverberated) and processes it with the sound of an actual or virtual acoustic space to create the illusion that the input was recorded in that space. The sound of the acoustic space is captured in what is called an impulse response (IR), which often starts as a recording of a short, sharp sound, such as the firing of a starter pistol or the bursting of an inflated balloon (the impulse), in the acoustic space in question. As you can imagine, such a sound excites the reverberation (the response) in the space, and so the impulse response (or at least its initial recording) sounds like an explosion followed by the reverberating reflections created by the recording space.
Once you have the IR for the space as a file, you then load it into the convolution reverb and input your sound to be processed. At that point, the software convolves the two digital audio signals together to create the output. Convolution itself is a mathematical process that has many applications including statistics, image processing, and electrical engineering as well as audio processing (It’s a very complex subject, understanding the precise details of how it works is not easy!). If you like, you can think of convolution as a kind of multiplication of each sample of the input with each sample of the IR, with the result being that the audio input takes on the sonic characteristics of the space in which the original IR was recorded.

Shimmer Reverb.
Shimmer reverb alters the pitch of the reverberated sound up or down in frequency by placing a pitch shifter in the delay lines feedback loop.
Once we apply any feedback in the reverb the pitch of the reverberated sound will steadily change up or down in pitch depending on the amount of pitch change and the feedback level. Shimmer reverb is often used in ambient music. Brian Eno has been one of the earliest adopters of this sound.

Creating a Flanger effect in SynthEdit.

03/12/2023 / Stephen / 0 Comments

What is a Flanger?

The term Flanger was coined sometime in the mid 20th century, It was discovered when two tape machines played the same signal at the same time. The story is that a recording engineer placed a finger on a tape reel’s flange, slowing one tape and throwing the two out of sync. When the engineer released the reel, its speed gradually returned to normal, giving a psychedelic whoosh. The origins of the term “Flanger” has various explanations…
How a Flanger works.
Flanging is an audio effect produced by mixing two identical signals together, one of the signals is delayed by a small and gradually changing period, usually between 5, and 20 milliseconds. This produces a swept comb filter effect: peaks and notches are produced in the resulting frequency spectrum, related to each other in a linear harmonic series.

Varying the time delay causes these to sweep up and down the frequency spectrum. A Flanger is an effects unit that creates this effect. Part of the output signal is usually fed back to the input (a “re-circulating delay line”), producing a resonance effect which further enhances the intensity of the peaks and troughs. The phase of the fed-back signal is sometimes inverted, producing another variation on the Flanger sound, the phase inversion causes peaks in the frequency response instead of notches (see the red line in the chart below).

How Positive and negative feedback affects the spectrum

As you can see Flanging differs from phasing in a notable way: A phaser generally has one notch frequency, whereas a Flanger has a number of harmonically related notch frequencies giving a comb like spectrum.

Creating a basic Flanger in SynthEdit.

The easiest way is to use a Delay2 module. Below is the structure of a simple Flanger. I find it’s best to keep the Mod depth’s maximum value to about 9 Volts as the full 10 Volts can sometimes produce some odd artefacts over 9.5 volts, we do this by setting the maximum value of the Mod Depth Slider control to 9 Volts.. The offset 5V is to keep the delay time in the centre of the useful delay time “window”.
Note: When constructing a Flanger in SE you should check the Interpolate box in the properties panel for the Delay2 module, and make sure it’s ticked. This greatly reduces any “stepping” or “zipper” effects in the audio when we are modulating the Delay2 module.

Adding an external Feedback loop with switchable phase.
With a few extra modules, we can create an external feedback circuit to give you a choice of negative or positive feedback loop for the Flanger.
Normally feedback loops are not allowed in SynthEdit, but if we add a special Feedback module we can have feedback. It does however introduce a delay of approximately 2mS into the loop however, which will affect the response of the filter above 2 kHz. See the structure below for the layout (the Feedback – Audio Module is actually Feedback – Volts Module, but in this context I think saying audio makes much more sense). The Phase Inverter is the standard Inverter2 Module, which just inverts the polarity of the audio, which is in effect a 180 degree phase change. The Feedback amp is a standard Level Adj Module and we use a Switch (1->Many) with a List Entry Module to switch from 0 degrees feedback to 180 degree phase shifted feedback. The feedback plug in this structure is not connected, and can be set at 0 Volts in the properties panel, or if you wish you can use a Fixed Value (Volts) module to hold the feedback level at 0 Volts.

Flanger with external feedback and phase switching.

Improving the SynthEdit Flanger.
Flangers may often produce a loud low frequency hum at high feedback levels. By adding a high-pass filter after the delay line we can fix this
problem.

Adding a High Pass filter in the output.
Here I have used two single pole filters to give a reasonable 12dB/Octave cut-off slope. You could have a fixed filter frequency, but I have used a slider control to allow a little manual control between 50 and 800Hz. To do this set both the 1 Pole HP filters Frequency Scales to 1V/kHz in the properties panel, and then set the Slider control’s Minimum to 0.05 Volts and the Maximum to 0.8 Volts.
There is a reason for this variable high pass filter, often we can improve the clarity of the Flanged signal by cutting out low frequencies so it’s nice to be able to control the cut-off frequency, and it can also remove some low frequency aliasing by-products.

Flanger with Phase switching and a High Pass Filter.

Making a stereo Flanger in SynthEdit.

Its quite a simple job to convert the simple Flanger to stereo.
We just need to add an extra input and output channel, so we have left and right sides, then copy parts of the basic Flanger setup: see below.
A nice touch is to change the LFO to the TD_LFO_B_ST module.
This is a “Stereo” version of the LFO we used previously, with two outputs instead of one, these outputs are also adjustable in terms of their phase relative to each other, so we could set our Left hand Flanger channel sweep voltage to be 180 degrees out of phase with the Right hand channel. Having the two channels sweeping differently will give us not only a more complex flanging effect, but also an interesting psych-acoustic panning effect too due to the phase and time differences between the two channels.

Month: December 2023

Ping-Pong (Cross) Echo.

Creating our Ping-Pong echo in SynthEdit.

Delay in SynthEdit.