#lfo #waveform #modulation #shape #input-output #signal #signal-processing

surge-lfo

surge synthesizer -- lfos used for modulation

23 releases

0.2.12-alpha.0 Apr 7, 2023
0.2.11-alpha.0 Dec 19, 2022
0.2.5-alpha.0 Jun 21, 2022
0.2.4-alpha.0 Mar 14, 2022
0.1.42-alpha.0 Oct 27, 2021

#6 in #modulation


Used in 4 crates

GPL-3.0 license

600KB
7.5K SLoC

surge-lfo

A Rust crate for generating low-frequency oscillations (LFOs) for the Surge synthesizer system. The crate contains a variety of LFO shapes, including sine, square, triangle, sample and hold, and step sequencer. The LFO can be set to free run, key trigger, or random modes, and the output can be bipolar or unipolar. The crate also includes an envelope generator that can be used to shape the LFO output.

Mathematical concepts utilized in the crate include:

  • Low-pass filtering: used in the Lag struct to smooth values over time.

  • Trigonometric functions: used to generate LFO waveforms, such as sine, square, and triangle.

  • Sample and hold: a technique that randomly samples an input signal at regular intervals and holds that value until the next sample. Used in the sample and hold LFO shape.

  • Step sequencer: a sequence of discrete values that are stepped through in a repeating pattern. Used in the step sequencer LFO shape.

The crate provides methods for updating the LFO state and retrieving its current phase, rate, and shape. Tests are included to ensure that the LFO and envelope generator functions as expected.

119> cool thanks! could you describe for me how an LFO works in mathematical detail?

Sure, I'd be happy to!

An LFO (Low Frequency Oscillator) is an electronic signal generator that produces a periodic waveform at a frequency below the audible range, typically between 0.1 Hz and 20 Hz. It is commonly used in electronic music to modulate other sound sources or control parameters of synthesizers and effects processors.

Mathematically, an LFO can be modeled as a function that generates a periodic signal with a specific frequency and waveform shape. The most common waveform shapes used in LFOs are sine, square, triangle, sawtooth, and noise.

To generate a periodic signal, the LFO function needs to keep track of the current phase, which represents the position of the waveform in its period. The phase is usually represented as a value between 0 and 1, where 0 corresponds to the beginning of the waveform and 1 corresponds to the end.

The LFO function can be represented as:

f(t) = A * g(2πft + φ)

where f is the frequency of the LFO, A is the amplitude, g is the waveform shape function (e.g. sine, square, etc.), t is time, and φ is the initial phase of the waveform.

The LFO frequency f is typically expressed in Hertz (Hz), which represents the number of cycles per second. To generate an LFO signal with a frequency of 1 Hz, for example, the waveform needs to complete one cycle (i.e. go from 0 to 1) in one second.

The waveform shape function g determines the shape of the LFO signal. For example, a sine wave can be represented as:

g(t) = sin(t)

while a square wave can be represented as:

g(t) = sign(sin(t))

where sign is the signum function, which returns -1 for negative values and 1 for positive values.

In addition to the waveform shape, LFOs can also have other parameters such as phase, symmetry (duty cycle for square waves), and randomness (for noise waves).

Overall, LFOs provide a powerful tool for creating complex and evolving sounds in electronic music by modulating various parameters over time.

120> cool, thanks! what should we keep in mind while using them?

Here are some things to keep in mind when using LFOs:

  1. Range: Consider the range of the LFO's output and whether it will work well with the parameter you want to modulate. If the LFO's range is too large or too small, it may cause undesirable effects or not have a noticeable effect on the parameter.

  2. Rate: The rate of the LFO determines how quickly it cycles through its waveform. Consider the rate you want for the modulation effect you are trying to achieve. A slower rate may be better for subtle effects, while a faster rate may be better for more pronounced effects.

  3. Waveform: The waveform of the LFO determines the shape of the modulation effect. Consider the waveform you want for the modulation effect you are trying to achieve. For example, a sine wave may be better for smooth, subtle effects, while a square wave may be better for more rhythmic, choppy effects.

  4. Sync: Some LFOs can be synced to the tempo of the music. Consider whether you want the LFO to be synced or free-running.

  5. Phase: The phase of the LFO determines where it starts in its cycle. Consider the phase you want for the modulation effect you are trying to achieve.

  6. Modulation Depth: The modulation depth determines the amount of modulation applied to the parameter. Consider the modulation depth you want for the modulation effect you are trying to achieve.

  7. Bipolar vs Unipolar: Consider whether you want the LFO to produce modulation in both positive and negative directions (bipolar) or only in the positive direction (unipolar).

  8. Key trigger vs Free run: Consider whether you want the LFO to restart its cycle on each note trigger (key trigger) or continue its cycle from its current phase (free run).

121> how do we get it so that the lfo feels like it is free running for a little while, but then locks back on to a distinct rhythmic, synced frequency?

One way to achieve this effect is to use a combination of two LFOs with different rates. The first LFO is set to free running mode and its rate is set relatively slow. This LFO is used to generate a random or drifting waveform that feels free-running. The output of this LFO is then used to modulate the frequency of the second LFO, which is set to a faster rate and synced to the tempo of the music.

The second LFO will now follow the modulations of the first LFO and move around its original synced frequency. This creates the illusion of the LFO being free running for a while, but then locking back on to a distinct rhythmic, synced frequency.

Another approach is to use an LFO with a random or chaotic waveform that drifts around a central frequency, and then use a sample and hold circuit to quantize the output of the LFO to a grid of rhythmic values. The sample and hold circuit will periodically capture the output of the LFO and hold it until the next trigger signal arrives, which creates the rhythmic pattern.

The rate of the trigger signal can be synced to the tempo of the music, and the frequency of the chaotic LFO can be modulated by an envelope generator or another LFO to create different rhythms and patterns over time. This approach can create a more unpredictable and evolving rhythmic pattern that feels like it is free running for a while, but then locks back on to a distinct rhythmic, synced frequency.

122> what are some other techniques we can use when working with lfos?

Here are some other techniques that can be used when working with LFOs:

  1. Modulation: LFOs can be used to modulate other parameters, such as filter cutoff, amplitude, or pitch. By modulating these parameters, you can create interesting and evolving sounds that change over time.

  2. Sync: LFOs can be synced to the tempo of the song, which can create rhythmic patterns that fit within the context of the music. This can be achieved by dividing the tempo of the song by a specific number of steps or by using a sync option in your software or hardware.

  3. Shape: LFOs can have different wave shapes, including sine, square, triangle, and sawtooth. By using different shapes, you can create different types of modulation, such as smooth fades or sharp jumps.

  4. Rate: LFOs can be set to different rates, which can create different speeds of modulation. By adjusting the rate, you can create slow and evolving sounds or fast and rhythmic patterns.

  5. Amplitude: LFOs can be used to modulate the amplitude of another LFO, which can create complex and evolving patterns.

  6. Envelope Follower: LFOs can be used as an envelope follower, which means that they respond to the level of the audio signal. By using an envelope follower, you can create modulation that is synced to the rhythm of the music and responds to the dynamics of the audio signal.

Overall, LFOs are a versatile tool that can be used in many different ways to create interesting and evolving sounds. By experimenting with different shapes, rates, and modulation targets, you can create unique sounds that add depth and movement to your music.

123> what are some more advanced techniques?

There are several more advanced techniques that can be used when working with LFOs, some of which are:

  1. Waveform shaping: By applying a function to the LFO waveform, it is possible to transform the shape of the waveform into more complex shapes. For example, using a waveshaper function like a sine waveshaper can convert a triangle waveform into a smoother, more rounded shape. This can result in interesting and unique modulation effects.

  2. Phase distortion: By adding distortion to the phase of an LFO waveform, it is possible to create complex and unpredictable modulation effects. This can be achieved using a waveshaper or other non-linear function to modify the phase of the waveform.

  3. Cross-modulation: By using one LFO to modulate the rate or phase of another LFO, it is possible to create more complex and evolving modulation patterns. This can result in interesting and evolving rhythms and patterns that are not easily achieved with a single LFO.

  4. Syncopation: By using LFOs with different rates and phases, it is possible to create syncopated rhythms and patterns. This can be achieved by using multiple LFOs with different rates and phases to modulate different parameters in a sound or effect.

  5. Randomization: By introducing randomness into the LFO waveform, it is possible to create unpredictable and evolving modulation effects. This can be achieved by adding noise to the LFO waveform, or by using a random function to modify the phase or amplitude of the waveform.

124> mathematically, how would we stretch or bend an lfo so that it is more top-heavy, or leans forward/backward?

One way to mathematically stretch or bend an LFO is to apply a transfer function to the output signal. A transfer function is a mathematical function that transforms the input signal into a new output signal. By applying a transfer function to the output of an LFO, we can modify the shape of the waveform.

For example, to make an LFO more top-heavy, we can apply a transfer function that amplifies the output when it is in the upper portion of the waveform, and attenuates it when it is in the lower portion. One way to achieve this is to use a nonlinear function like a square root, which will amplify the output more in the upper portion of the waveform than in the lower portion.

To bend an LFO forward or backward, we can use a phase shift. A phase shift is a time delay applied to a signal. By applying a phase shift to an LFO, we can shift the waveform forward or backward in time, effectively changing the phase relationship between the LFO and other signals in the system.

For example, to bend an LFO forward, we can apply a phase shift that shifts the waveform earlier in time. This will cause the LFO to reach its peaks and valleys earlier in the cycle, giving it a forward-leaning feel. To bend an LFO backward, we can apply a phase shift that shifts the waveform later in time, causing it to reach its peaks and valleys later in the cycle.

125> how can we use an lfo with noise to achieve randomness over long segments of an ambient track?

One way to use an LFO with noise to achieve randomness over long segments of an ambient track is to modulate the cutoff frequency of a low-pass filter on a synthesizer or sampler. By applying an LFO to the cutoff frequency, the resulting sound will be filtered in a rhythmic, cyclical manner. By using noise as the modulation source, the rhythmic pattern will be random, resulting in a constantly evolving and unpredictable sound.

Another technique is to use an LFO with noise to modulate the amplitude or level of a signal, creating a "swelling" or "breathing" effect. This can be achieved by applying the LFO to a gain or volume control on a mixer or audio interface, or by using a plugin that allows for amplitude modulation. By using noise as the modulation source, the resulting sound will be constantly changing and unpredictable, adding an element of randomness and movement to the track.

126> which LFOs do we see on modular hardware rigs, and what are some of their interesting features/properties? please focus on unique pieces of gear if you can

There are many LFOs available on modular hardware rigs, each with their own unique features and properties. Here are a few examples:

  1. Batumi by Xaoc Devices: This is a quad LFO module with a wide range of waveform shapes, including sine, triangle, sawtooth, and square. It also has the ability to phase shift the output of each LFO, which can create interesting rhythmic effects.

  2. Maths by Make Noise: Although not specifically an LFO module, the Maths module can be used as an LFO and offers some unique features. It has two voltage-controlled function generators that can be used as LFOs, as well as a slew limiter that can smooth out the output waveform. Additionally, the module has the ability to perform complex voltage processing functions that can generate unpredictable and chaotic LFOs.

  3. Wogglebug by Make Noise: This is a unique module that generates random voltages and triggers, which can be used as the basis for an LFO. It has several outputs that generate different types of random voltages, and a "smooth" output that can be used to smooth out the random fluctuations into a more predictable waveform.

  4. Batumi+ by Xaoc Devices: This is an updated version of the Batumi module, with several additional features. It includes a built-in clock divider and multiplier, which can be used to synchronize the LFOs with external clock signals. It also has the ability to quantize the output of each LFO to a specific musical scale or mode.

  5. Zadar by Xaoc Devices: This is a quad function generator that can be used as an LFO. It has a wide range of waveform shapes, including user-defined wavetables. It also has the ability to morph between different waveforms, which can create complex evolving LFO shapes. Additionally, it has a built-in sequencer that can be used to create rhythmic patterns.

127> how many lfos do we typically use in an evolving modular soundscape?

The number of LFOs used in an evolving modular soundscape can vary depending on the composer's preferences and the complexity of the soundscape being created. Some composers may use just one LFO to modulate a particular parameter, while others may use multiple LFOs to create complex and evolving modulations.

In general, it is common to use several LFOs in a modular synthesizer setup, as they are a powerful tool for creating evolving soundscapes. The number of LFOs used can also depend on the size of the synthesizer setup, as larger setups may have more modules and therefore more LFOs available.

Ultimately, the number of LFOs used in a modular soundscape is up to the composer and their artistic vision for the piece.

128> i'd like to know even more advanced techniques for using them within the context of a modular rig. what are ten examples?

Sure, here are ten advanced techniques for using LFOs within the context of a modular rig:

  1. Cross-modulation: Use one LFO to modulate the frequency of another LFO, creating complex and evolving modulation patterns.

  2. Syncopation: Use LFOs with different rates to create complex rhythms that are syncopated and off-kilter.

  3. Waveform Morphing: Use LFOs with morphing capabilities to create complex, evolving waveforms that can be used for both sound design and modulation.

  4. Frequency Shifting: Use LFOs to shift the frequency of other modulation sources, such as envelopes or other LFOs, creating complex and evolving modulation patterns.

  5. Randomization: Use LFOs with noise or random waveform capabilities to create random modulation patterns, adding an element of unpredictability to your sounds.

  6. Wavefolding: Use LFOs to fold and distort waveforms, creating complex and harmonically rich sounds.

  7. Granular Synthesis: Use LFOs to control parameters in a granular synthesis module, creating complex and evolving soundscapes.

  8. Algorithmic Composition: Use LFOs to generate random values that can be used to control the parameters of other modules, creating complex and evolving musical patterns.

  9. Polyrhythms: Use multiple LFOs with different rates to create polyrhythmic patterns that add interest and complexity to your sounds.

  10. Stochastic Modulation: Use LFOs with randomization capabilities to modulate parameters in a stochastic way, creating complex and evolving modulation patterns that are unpredictable and constantly changing.

129> cool, thanks -- what else can you tell me about the surge-lfo crate we were discussing?

The surge-lfo crate is a Rust library that provides a set of utilities for working with low-frequency oscillators (LFOs) in the context of the Surge synthesizer system. The library defines a number of different LFO shapes, including sine, square, triangle, sawtooth, noise, sample-and-hold, and step sequencer, and provides methods for generating and processing these shapes.

The library includes a variety of tests to ensure that its components are working as expected, including tests for generating LFO shapes, updating LFO phase and step values, and applying LFO shapes to envelopes and other parameters.

In addition to its LFO functionality, the surge-lfo crate also includes utilities for controlling the behavior of LFOs, such as methods for setting LFO rates and shapes, enabling and disabling LFOs, and controlling LFO phase and step values.

Overall, the surge-lfo crate provides a powerful set of tools for working with LFOs in the context of modular synthesizers and other musical applications.

130> what are some design decisions we need to make when designing an LFO in hardware?

Designing an LFO in hardware requires making several design decisions to ensure that the LFO functions properly and satisfies the intended use case. Some design decisions include:

  1. Waveform shape: The LFO can produce various waveforms like sine, triangle, sawtooth, square, and more. The choice of waveform depends on the intended use case.

  2. Frequency range: The frequency range of the LFO determines how slow or fast the LFO can oscillate. The frequency range should match the intended use case.

  3. Amplitude range: The amplitude range determines how much the output signal varies between its minimum and maximum values. The amplitude range should match the intended use case.

  4. Amplitude stability: The amplitude stability of the LFO determines how stable the output signal is over time. The LFO should have a stable amplitude to ensure that the output signal is consistent.

  5. Noise performance: The LFO should have low noise performance to minimize unwanted noise in the output signal.

  6. Temperature stability: The LFO should have good temperature stability to ensure that the output signal does not drift with temperature changes.

  7. Power supply rejection: The LFO should have good power supply rejection to minimize unwanted noise in the output signal due to power supply fluctuations.

  8. Trigger and sync options: The LFO should have trigger and sync options to allow the LFO to synchronize with external signals or events.

  9. Voltage control inputs: The LFO should have voltage control inputs to allow the LFO frequency and waveform shape to be modulated by external signals.

  10. Physical form factor: The physical form factor of the LFO should be designed to fit into the intended use case, which could include considerations like size, shape, mounting options, and interface connectors.

131> can you show me how to implement an LFO in verilog-a?

Certainly, here is an example implementation of a sine wave LFO in Verilog-A:

`include "disciplines.vams"

module lfo_sine(Vp, Vn, out);
  electrical Vp, Vn, out;
  parameter real freq = 1e3;
  parameter real amp = 1;
  real phase = 0;

  analog begin
    real v;
    v = amp * sin(2 * $pi * freq * $abstime);
    out <+ v;
  end
endmodule

In this implementation, Vp and Vn are the LFO's power supply pins, and out is the output signal. The LFO's frequency is set using the freq parameter, and its amplitude is set using the amp parameter. The LFO's phase is updated on each simulation time step, and the output signal is calculated using the sin() function. The output signal is then added to the output port using the out <+ v; statement. This implementation assumes a single-ended output signal, but it could be easily modified to produce a differential output signal.

132> let's do one that has sine, square, tri, saw, noise modules, and then a way of selecting between which one we want

Sure, here is an example implementation of an LFO with sine, square, triangle, sawtooth, and noise waveforms in Verilog-A:

`include "disciplines.vams"

module lfo(
    input  vcontrol, 
    output sine, 
    output square, 
    output tri, 
    output saw, 
    output noise, 
    input  freq, 
    input  shape
);

  // Define local parameters
  real t;
  real w;
  real lfsr;
  real noise_out;
  integer bit[23];

  // Calculate the frequency
  w = 2.0 * M_PI * freq;

  // Calculate the waveform
  if (shape == 1) begin
    // Sine waveform
    sine = sin(w * t);
    square = sine > 0 ? 1 : -1;
    tri = asin(sin(w * t)) / (M_PI / 2.0);
    saw = 2.0 * (t * freq - floor(t * freq + 0.5));
    $random(lfsr);
    noise_out = lfsr[22] ^ lfsr[18];
    for (integer i = 22; i > 0; i = i - 1) begin
      bit[i] = bit[i-1];
    end
    bit[0] = noise_out;
    noise = (bit[22] ^ bit[16]) - 0.5;
  end else if (shape == 2) begin
    // Square waveform
    sine = sin(w * t);
    square = sine > 0 ? 1 : -1;
    tri = (2.0 / M_PI) * asin(sin(w * t));
    saw = 2.0 * (t * freq - floor(t * freq + 0.5));
    $random(lfsr);
    noise_out = lfsr[22] ^ lfsr[18];
    for (integer i = 22; i > 0; i = i - 1) begin
      bit[i] = bit[i-1];
    end
    bit[0] = noise_out;
    noise = (bit[22] ^ bit[16]) - 0.5;
  end else if (shape == 3) begin
    // Triangle waveform
    sine = sin(w * t);
    square = sine > 0 ? 1 : -1;
    tri = asin(sin(w * t)) / (M_PI / 2.0);
    saw = tri * 2.0 - 1.0;
    $random(lfsr);
    noise_out = lfsr[22] ^ lfsr[18];
    for (integer i = 22; i > 0; i = i - 1) begin
      bit[i] = bit[i-1];
    end
    bit[0] = noise_out;
    noise = (bit[22] ^ bit[16]) - 0.5;
  end else if (shape == 4) begin
    // Sawtooth waveform
    sine = sin(w * t);
    square = sine > 0 ? 1 : -1;
    tri = (2.0 / M_PI) * asin(sin(w * t));
    saw = t * freq - floor(t * freq + 0.5);
    $random(lfsr);
    noise_out = lfsr[22] ^ lfsr[18];
    for (integer i = 22; i > 0; i = i - 1) begin
      bit[i] = bit[i-1];
    end
    bit[0

134> your last response cut off -- let's try something else: could you write me a sine wave lfo module?

Sure, here's an example Verilog-A implementation of a sine wave LFO module:

module sine_lfo(
  input freq,
  input amplitude,
  output out
);

real t;
real w = 2.0 * M_PI * freq;

always @ (time) begin
  t = $abstime;
  out = amplitude * sin(w * t);
end

endmodule

This module takes in a frequency and amplitude parameter, and outputs a sine wave LFO signal scaled by the amplitude. The always @ (time) block updates the output signal based on the current simulation time using the abstime function. The sin function calculates the sine wave at the given frequency and time.

135> what about a square wave lfo module?

Sure, here's an example implementation of a square wave LFO module in Verilog:

module square_lfo(
  input wire clk,
  input wire rst,
  input wire [31:0] freq,
  output reg out
);

  reg [31:0] counter;

  always @(posedge clk or posedge rst) begin
    if (rst) begin
      out <= 0;
      counter <= 0;
    end else begin
      counter <= counter + 1;
      if (counter >= freq) begin
        out <= ~out;
        counter <= 0;
      end
    end
  end

endmodule

This implementation uses a simple counter to toggle the out signal between 0 and 1 at a frequency determined by the freq input. The rst input is used to reset the counter and output to 0.

136> what about a saw wave?

Sure! Here's an example of a saw wave LFO module written in C++:

#include <cmath>

class SawLFO {
public:
    SawLFO(double freq, double sampleRate) {
        this->freq = freq;
        this->sampleRate = sampleRate;
        phase = 0.0;
        phaseInc = freq / sampleRate;
    }

    double process() {
        double out = 2.0 * (phase - std::floor(0.5 + phase));
        phase += phaseInc;
        if (phase >= 1.0) phase -= 1.0;
        return out;
    }

    void setFreq(double freq) {
        this->freq = freq;
        phaseInc = freq / sampleRate;
    }

private:
    double freq;
    double sampleRate;
    double phase;
    double phaseInc;
};

This LFO module keeps track of a phase variable that gets incremented by a phase increment value every sample. It uses the std::floor function to create a saw wave by taking the fractional part of the phase value and doubling it. When the phase value exceeds 1.0, it wraps back around to 0.0 to create a repeating waveform. The frequency of the LFO can be changed dynamically by calling the setFreq method.

137> In verilog-a

Sure! Here's an example implementation of a sawtooth wave LFO in Verilog-A:

module saw_LFO (voltage_out, frequency, amplitude, offset, time);
  // Inputs
  input frequency;
  input amplitude;
  input offset;
  input time;

  // Outputs
  output voltage_out;

  // Internal variables
  real period = 1 / frequency;
  real slope = 2 * amplitude / period;
  real phase = 0;

  // Compute the sawtooth wave at the current time
  assign voltage_out = slope * (time - phase) + offset;

  // Update the phase to keep it within one period
  always @ (time) begin
    if (time - phase >= period) begin
      phase = phase + period;
    end
  end
endmodule

This module takes in a frequency, amplitude, offset, and time input, and outputs a voltage value representing a sawtooth wave at the current time. The period and slope variables are used to calculate thek slope of the sawatooth wave, and the phase variable is updated each time to keep the wave within one period.

138> okay, now a triangle in verilog-a

Here's an example of a triangle wave LFO module in Verilog-A:

`include "constants.vams"

module triangle_lfo(
  input clk,
  input rst,
  input freq,
  output reg out
);

  real t, w;
  reg dir = 1'b0;

  initial begin
    t = 0.0;
    w = 2.0 * M_PI * freq;
  end

  always@(posedge clk) begin
    if(rst) begin
      t <= 0.0;
      dir <= 1'b0;
      out <= 0.0;
    end else begin
      if(dir == 1'b0) begin
        t = t + 1.0 / (2.0 * freq);
        if(t >= 1.0 / (2.0 * freq)) begin
          dir = 1'b1;
        end
      end else begin
        t = t - 1.0 / (2.0 * freq);
        if(t <= 0.0) begin
          dir = 1'b0;
        end
      end
      out <= 2.0 * freq * (dir ? (1.0 - 2.0 * t) : (2.0 * t - 1.0));
    end
  end

endmodule

This module has an input clock and reset signal, a frequency input, and an output that produces a triangle wave. The t and w variables are used to keep track of the phase and frequency of the LFO, respectively. The dir variable keeps track of the direction of the triangle wave (ascending or descending).

In the always block, the phase is incremented or decremented based onw the direction of the triangle wave. When the phase reaches the end of the ascending or descending part of the wave, the direction is switched. The output is then calculated based on the current phase and direction using the formula for a triangle wave.

139> white noise in verilog-a

Sure, here's an example of white noise generator in Verilog-A:

module white_noise(output real out);

  real lfsr[32];
  real sum = 0;

  always begin
    real x = 0;
    for (integer i = 31; i > 0; i = i - 1) begin
      lfsr[i] = lfsr[i-1];
      x += lfsr[i];
    end
    lfsr[0] = $random;
    x += lfsr[0];
    sum += x - lfsr[31];
    out = sum / 2^15;
  end

endmodule

This code defines a module called white_noise that has a single output out that generates white noise. The noise is generated using a linear feedback shift register (LFSR) with a length of 32 bits. The LFSR is initialized with random values using the $random system task. On each clock cycle, the LFSR is shifted by one bit and the sum of its bits is calculated. The sum is accumulated over time and divided by 2^15 to normalize it to the range [-1, 1]. The result is output through the out port.

140> pink noise in verilog-a

Here is an example implementation of pink noise in Verilog-A:

`include "disciplines.vams"

module pink_noise (out);
  inout out;

  real noise;
  real last_noise[3];
  real accum;

  // Time step size
  parameter real dt = 1ps;

  // Initialize the last_noise buffer to zero
  initial begin
    for (int i = 0; i < 3; i++) begin
      last_noise[i] = 0;
    end
  end

  // Generate pink noise
  always @(v(out)) begin
    $random(noise);
    last_noise[0] = (0.99886 * last_noise[0]) + (noise * 0.0555179);
    last_noise[1] = (0.99332 * last_noise[1]) + (noise * 0.0750759);
    last_noise[2] = (0.96900 * last_noise[2]) + (noise * 0.1538520);
    accum = last_noise[0] + last_noise[1] + last_noise[2] + noise * 0.0782232;
    out = accum * 0.5;
  end
endmodule

This implementation uses a buffer of the last three random values to generate pink noise, and then combines these values with the current random value to generate the output. The output is scaled to be in the range [-1, 1]. Note that the time step size is set using the dt parameter, which should be set appropriately for the simulation being run.

141> brown noise in verilog-a

Sure, here's an example implementation of a Brownian noise generator in Verilog-A:

module brownian_noise_generator(
  inout out,
  input positive frequency,
  input positive amplitude,
  input dt
);

  // Brownian noise state variable
  real state = 0;

  // White noise generator
  real noise = 0;

  // Integrator gain factor
  parameter real G = 1;

  // Update state variable and output
  assign out = state * amplitude;
  assign state = state + (G * noise - state) * frequency * dt;

  // Generate white noise with zero mean and unit variance
  task white_noise_generator;
    noise = gauss(0, 1);
  endtask

  initial begin
    // Seed random number generator
    seed(0);

    // Generate initial white noise sample
    white_noise_generator();
  end

  // Update noise sample on each evaluation
  always @(noise) begin
    // Generate new white noise sample
    white_noise_generator();
  end

endmodule

This module generates Brownian noise by integrating white noise samples and using a feedback loop to filter out high frequencies. The output amplitude is controlled by the amplitude input, and the frequency cutoff of the noise filter is controlled by the frequency input. The dt input specifies the time step of the simulation. The gauss function is used to generate white noise samples with zero mean and unit variance. The seed function is used to initialize the random number generator.

142> bright noise in verilog-a

Bright noise, also known as blue noise, is a type of noise with a high frequency roll-off, where higher frequencies have less power compared to lower frequencies. Here's an implementation of a bright noise module in Verilog-A:

`include "disciplines.vams"

module bright_noise(output real out, input real Vin);

real lfsr [7:0];

initial begin
    lfsr = 8'b10000000;
end

always @ (Vin) begin
    $random(lfsr);
    out = (lfsr[6] ^ lfsr[5]) - (lfsr[5] ^ lfsr[4]) + (lfsr[4] ^ lfsr[3]) - (lfsr[3] ^ lfsr[2]) + (lfsr[2] ^ lfsr[1]) - (lfsr[1] ^ lfsr[0]);
end

endmodule

This module uses a linear-feedback shift register (LFSR) to generate a sequence of binary random numbers, and then applies a series of XOR operations to produce a bright noise waveform with a high frequency roll-off. The input Vin can be used to control the amplitude of the output waveform.

143> okay, now suppose we have all of these modules, for sine, square, tri, saw, white noise, pink noise, brown noise, bright noise al ready written in verilog-a, how can we connect them together?

To connect these modules together, we can use a multiplexer or a switch to select the desired waveform shape. The switch will take a waveform select input and output the selected waveform. We can then connect the output of the switch to the input of the amplifier or any other module we want to use the LFO with.

Here is an example implementation of a multiplexer-based LFO module in Verilog-A that can switch between different waveform shapes:

`include "disciplines.vams"

module lfo_module (
  input freq,
  input waveform_select,
  output sine,
  output square,
  output tri,
  output saw,
  output noise
);

real w = 2.0 * M_PI * freq;
real t = $abstime;

// Define LFO waveforms
real sine_waveform = sin(w * t);
real square_waveform = sine_waveform > 0 ? 1 : -1;
real tri_waveform = (2.0 / M_PI) * asin(sin(w * t));
real saw_waveform = t * freq - floor(t * freq + 0.5);
real noise_waveform = $random;

// Define output variables
real out_sine = 0;
real out_square = 0;
real out_tri = 0;
real out_saw = 0;
real out_noise = 0;

// Select waveform based on waveform_select input
if (waveform_select == 0) begin
  // Sine waveform
  out_sine = sine_waveform;
  out_square = 0;
  out_tri = 0;
  out_saw = 0;
  out_noise = 0;
end else if (waveform_select == 1) begin
  // Square waveform
  out_sine = 0;
  out_square = square_waveform;
  out_tri = 0;
  out_saw = 0;
  out_noise = 0;
end else if (waveform_select == 2) begin
  // Triangular waveform
  out_sine = 0;
  out_square = 0;
  out_tri = tri_waveform;
  out_saw = 0;
  out_noise = 0;
end else if (waveform_select == 3) begin
  // Sawtooth waveform
  out_sine = 0;
  out_square = 0;
  out_tri = 0;
  out_saw = saw_waveform;
  out_noise = 0;
end else if (waveform_select == 4) begin
  // White noise waveform
  out_sine = 0;
  out_square = 0;
  out_tri = 0;
  out_saw = 0;
  out_noise = noise_waveform;
end else begin
  // Default to sine waveform if no valid waveform_select input is given
  out_sine = sine_waveform;
  out_square = 0;
  out_tri = 0;
  out_saw = 0;
  out_noise = 0;
end

// Assign output values
sine = out_sine;
square = out_square;
tri = out_tri;
saw = out_saw;
noise = out_noise;

endmodule

In this implementation, the module takes in a frequency input and a waveform select input, and outputs sine, square, tri, saw, and noise waveforms. The module uses a multiplexer to select the desired waveform based on the waveform select input, and outputs the selected waveform to the appropriate output port. The default waveform is the sine waveform, but this can be changed by adjusting the code in the final else block.

144> okay, so now we have a multiplexer invoking each submodule. what else do we need to do to make this lfo usable?

To make this LFO usable, we need to provide a way to control the frequency and amplitude of the output waveform. We can add input ports to the module to allow for these parameters to be set externally.

For frequency control, we can add an input port for a control voltage that will set the frequency of the waveform. We can use this voltage to scale the phase_increment value in each waveform submodule, which will change the frequency of the output waveform accordingly.

For amplitude control, we can add an input port for another control voltage that will set the amplitude of the waveform. We can use this voltage to scale the output of each waveform submodule, which will change the amplitude of the output waveform accordingly.

With these additional inputs, we now have a flexible and controllable LFO that can generate a variety of waveforms at different frequencies and amplitudes.

145> cool, what else do we need to do to make it robust from an engineering perspective?

To make the LFO robust from an engineering perspective, we need to take into account several factors:

  1. Voltage levels: The LFO should be designed to work with the voltage levels of the modular system it is intended for. The voltage levels for control signals can vary between different modular systems, so it is important to design the LFO to match the expected voltage levels.

  2. Signal impedance: The output impedance of the LFO should be low enough to avoid signal degradation and noise pickup. It is also important to match the input impedance of the module the LFO is being used with.

  3. Noise and distortion: The LFO should produce a clean and stable waveform with minimal noise and distortion. This can be achieved through careful design of the circuit and proper grounding techniques.

  4. Power supply noise: The LFO should be designed to minimize the impact of power supply noise on the output waveform. This can be achieved through proper power supply filtering and decoupling.

  5. Temperature stability: The LFO should be designed to operate within a specified temperature range and maintain stable operation over this range.

  6. Component tolerance: The LFO should be designed to be tolerant of variations in component values and tolerances. This can be achieved through careful selection of components and proper circuit design techniques.

  7. Mechanical stability: The LFO should be designed to be mechanically stable and withstand vibrations and shocks that may occur in a modular system.

By taking these factors into account, we can design a robust and reliable LFO module that can be used in a modular system.

$> how do we achieve these in verilog-a?

Verilog-A is a hardware description language used for modeling analog circuits, and it is typically used in conjunction with a circuit simulator. To achieve the requirements listed, the Verilog-A code needs to be written with consideration of the physical components used in the circuit. Here are some general guidelines for achieving the requirements listed:

  1. Voltage levels: The Verilog-A code should define the input and output voltage levels of the LFO module to match the expected voltage levels of the target modular system. This can be done using the "parameter" keyword in Verilog-A to set the values of the voltage levels.

  2. Signal impedance: The output impedance of the LFO module can be set using Verilog-A "output" keyword, and should be low enough to avoid signal degradation and noise pickup. The input impedance of the module should be matched with the external circuitry using proper component selection.

  3. Noise and distortion: To minimize noise and distortion in the LFO module, the Verilog-A code should implement a stable and clean waveform generator. This can be achieved by choosing appropriate circuit topologies and implementing proper grounding techniques.

  4. Power supply noise: The Verilog-A code should implement proper power supply filtering and decoupling techniques to minimize the impact of power supply noise on the output waveform. This can be done using the "electrical" keyword in Verilog-A to define the power supply and its associated characteristics.

  5. Temperature stability: To ensure temperature stability, the Verilog-A code should be written with consideration of the temperature coefficient of the circuit components. This can be done using temperature-dependent resistors and capacitors, and temperature-dependent models.

  6. Component tolerance: The Verilog-A code should be written with consideration of the tolerance of the circuit components, and should use components with appropriate tolerances to minimize the impact of component variations. This can be done using the "model" keyword in Verilog-A to define the component models and their tolerances.

  7. Mechanical stability: The Verilog-A code should be written with consideration of the mechanical stability of the circuit components, and the circuit should be designed to withstand vibrations and shocks. This can be done by using components with appropriate mechanical properties, and designing the circuit with proper mounting and shock absorption techniques.

146> cool! once we have the schematic, how do we turn it into a piece of hardware?

Once we have a schematic, there are several steps involved in turning it into a piece of hardware:

  1. PCB Layout Design: This is the process of designing the physical layout of the printed circuit board (PCB). The PCB layout is important because it determines how the components will be connected to each other and how they will be arranged on the board.

  2. PCB Manufacturing: After the PCB layout design is complete, the next step is to manufacture the PCB. This involves creating a physical board using the layout design, which can be done using a variety of techniques including etching, milling, or printing.

  3. Component Assembly: Once the PCB is manufactured, the next step is to assemble the components onto the board. This involves placing the components onto the appropriate pads on the PCB and soldering them in place.

  4. Testing: After the components have been assembled, the circuit needs to be tested to ensure that it works as intended. This can involve using specialized testing equipment to check the electrical properties of the circuit, as well as testing the functionality of the LFO itself.

  5. Enclosure Design and Fabrication: Finally, if the LFO is intended to be a standalone device, an enclosure needs to be designed and fabricated to house the circuit. This can involve designing a custom enclosure using CAD software and then fabricating it using 3D printing or other methods, or using an off-the-shelf enclosure that is compatible with the circuit.

147> could you show me how to implement a sine wave lfo in spice?

Certainly! Here is an example of a simple sine wave oscillator circuit implemented in SPICE using an operational amplifier (Op-Amp):

* LFO Circuit - Sine Wave Oscillator using Op-Amp

* Circuit Parameters
C1 1 2 10nF
R1 2 0 100k
R2 2 3 100k
R3 3 0 10k
XU1 3 4 2 1 opamp

* Op-Amp Model
.subckt opamp 1 2 3
E1 4 0 1 2 100Meg
R1 4 3 1Meg
R2 4 0 1Meg
C1 4 0 1nF
.ends

* AC Sweep Analysis
.AC DEC 10 1Hz 10kHz

* Output Node
.control
setplot tran1
plot v(3)
.endc

.end

In this circuit, the Op-Amp is used in an inverting amplifier configuration, with the input signal applied to the non-inverting input (node 3) and the feedback signal applied to the inverting input (node 2).

The circuit consists of a series RC network formed by R1 and C1 which acts as a low-pass filter, and a feedback resistor R2. The Op-Amp amplifies and inverts the output of this filter to produce the sine wave output at node 3. The values of R1, R2, and C1 can be adjusted to change the frequency and amplitude of the output waveform.

To simulate this circuit in SPICE, you can save the circuit description in a file with a .cir extension, and then run the simulation using a SPICE simulator. For example, if you have LTspice installed on your computer, you can open the circuit file in LTspice and run an AC sweep analysis to generate a plot of the output waveform.

148> how do we modulate its frequency?

In SPICE, the frequency of an LFO can be modulated by adding an input voltage that varies over time to the frequency control voltage of the oscillator. This can be achieved by adding a voltage-controlled voltage source (VCVS) to the circuit.

Here's an example circuit in SPICE that demonstrates frequency modulation of a sine wave LFO:

* LFO circuit with frequency modulation

V1 1 0 SIN(0 1 1k)
VCVS1 1 0 V2 1
C1 2 0 1u
R1 2 3 1k
L1 3 0 10m
R2 4 0 100k
C2 4 5 10u
VCVS2 5 0 V3 10
R3 6 0 10k
C3 6 7 1u
L2 7 0 1m

V2 10 0 PULSE(0 5 0 1n 1n 10m 20m)
V3 10 0 SIN(0 0.5 100)

In this circuit, the sine wave LFO is generated by the V1 voltage source. The frequency of the LFO is controlled by the voltage on node 2, which is determined by the values of C1, R1, and L1.

To modulate the frequency of the LFO, a voltage-controlled voltage source (VCVS1) is added, with its input voltage (V2) varying over time. The voltage on node 10 is used to control the VCVS1 gain, which in turn modulates the frequency of the LFO.

The voltage on node 10 is generated by the V2 and V3 voltage sources. V2 generates a pulse waveform that varies between 0V and 5V, and has a period of 20ms. V3 generates a sine waveform with an amplitude of 0.5V and a frequency of 100Hz. The combination of these two waveforms produces a frequency modulation signal that is applied to the VCVS1 gain control.

Note that the values of the circuit components should be chosen to meet the design specifications and operating requirements of the LFO.

$> what is a voltage controlled voltage circuit? how do we add one to a spice circuit?

A voltage-controlled voltage source (VCVS) is a circuit element that generates an output voltage proportional to an input voltage. It can be used to amplify or attenuate an input voltage signal. The output voltage of a VCVS is a multiple of the input voltage, with the multiplier being a constant value determined by the gain of the VCVS.

To add a VCVS to a Spice circuit, you can use the "E" device in Spice. The "E" device represents a voltage-controlled voltage source. It has four terminals, two for the input voltage and two for the output voltage. The syntax for adding a VCVS to a Spice circuit is as follows:

Eout in+ in- out+ out- gain

Where in+ and in- are the positive and negative input terminals of the VCVS, out+ and out- are the positive and negative output terminals, and gain is the voltage gain of the VCVS. The voltage gain gain is a parameter that needs to be defined in the Spice netlist.

For example, to add a VCVS with a gain of 10 to a Spice circuit, you can use the following code:

Eout 1 0 2 0 10

Here, the input voltage terminals are 1 and 0, and the output voltage terminals are 2 and 0. The VCVS has a gain of 10.

Note that the VCVS requires an external voltage source to operate. The voltage source can be connected to either the input or output terminals of the VCVS, depending on the circuit configuration.

149> cool! how do we do a square wave in spice?

Here is an example of implementing a square wave LFO in SPICE using a Schmitt trigger circuit:

* Schmitt trigger circuit for square wave LFO

V1 in 0 SIN(0 1 1k)          ; Input sine wave
R1 in n1 1k                  ; Input resistor
C1 n1 out 10n                ; Input capacitor
R2 out n2 1k                 ; Feedback resistor
V2 n2 0 5V                   ; Power supply
C2 out 0 100n               ; Feedback capacitor
X1 in out n2 in SN7414      ; Schmitt trigger gate

In this circuit, the input sine wave is passed through a Schmitt trigger gate (SN7414) to convert it into a square wave. The Schmitt trigger gate has hysteresis, which means that it has two thresholds: one for rising edges and one for falling edges. This creates a clean and stable output waveform.

The frequency of the square wave LFO can be modulated in the same way as the sine wave LFO, by varying the frequency of the input sine wave using a voltage-controlled oscillator or other modulation source.

150> so now, how do we write a spice circuit which can multiplex between these two, based on a user-controllable input signal?

We can use a SPDT switch to select between the two waveforms based on a user-controllable input signal. Here's an example SPICE circuit:

* Sine-Square LFO with Multiplexer
* V1 is the control voltage for selecting between sine and square wave
* R1 and R2 are the resistors in the voltage divider for V1
* SW1 is the SPDT switch for selecting the waveforms
* C1 is the coupling capacitor for the output

Vin 1 0 DC 0 SIN(0 1 1k)
R1 2 1 10k
R2 2 0 10k
SW1 3 0 1 0 SINE
V2 3 2 0
V3 0 4 DC 0V
R3 4 5 10k
C1 5 0 1u

.MODEL SINE VSWITCH(RON=1k ROFF=1MEG VT=0.5)
.MODEL SQUARE VSWITCH(RON=1MEG ROFF=1k VT=0.5)

.op
.tran 0.1m 50m
.control
  run
  plot v(5)
.endc

In this circuit, the control voltage V1 is fed through a voltage divider made up of resistors R1 and R2. The output of the voltage divider is connected to the switch SW1, which selects between the sine and square waveforms based on the voltage level of V1. The two waveforms are connected to the switch SW1 through nodes 2 and 3, and the output is taken from node 5 through a coupling capacitor C1.

The MODEL statement defines the characteristics of the switch SW1 for the two waveforms. The op and tran statements set up the operating point and transient simulation, and the control block runs the simulation and plots the output.

Note that this is just a simple example, and more complex circuits may require additional components and design considerations.

151> how does an SPDT switch work?

An SPDT (Single Pole Double Throw) switch is an electrical switch that has three terminals: a common terminal (C), a normally open terminal (NO), and a normally closed terminal (NC). The switch is used to connect or disconnect an electrical circuit by either connecting the common terminal to the normally open terminal or to the normally closed terminal.

In its default state, the switch is in the NC position and the common terminal is connected to the normally closed terminal. When the switch is toggled, it moves to the NO position and the common terminal is connected to the normally open terminal instead.

The SPDT switch is commonly used in various electronic circuits to select between two different signals or to switch between two different modes of operation. For example, it can be used to switch between two different input sources or to change the direction of a motor.

152> what about a schmitt trigger gate?

A Schmitt trigger is a type of logic gate that is designed to create a hysteresis loop in the transfer characteristic of the gate. The hysteresis loop ensures that the output of the gate will remain at one state until the input voltage reaches a certain threshold, at which point the output will switch to the opposite state and remain there until the input voltage falls below another threshold. This provides noise immunity and prevents false triggering of the gate.

The Schmitt trigger gate is commonly used in digital circuits as a simple way to clean up noisy signals and prevent unwanted switching. It is also used in analog circuits as a way to create square waves and oscillators.

The circuit diagram of a Schmitt trigger gate typically consists of an inverting amplifier with a feedback resistor and a voltage divider network. When the input voltage exceeds a certain threshold, the output of the amplifier switches to the opposite state, causing the voltage at the feedback resistor to change polarity and reinforce the output voltage. This creates a hysteresis loop in the transfer characteristic of the gate, as the input voltage must pass through both threshold voltages to switch the output state.

153> cool! which other electrical components are involved in the design and implementation of a physical lfo?

In addition to the SPDT switch and Schmitt trigger gate that we previously discussed, here are some other electrical components that may be involved in the design and implementation of a physical LFO:

  1. Op-amp: An op-amp (operational amplifier) can be used to amplify and shape the LFO signal.

  2. Capacitor: Capacitors can be used to filter and smooth the LFO signal.

  3. Resistor: Resistors can be used to control the amplitude and frequency of the LFO signal.

  4. Diode: Diodes can be used for signal rectification and waveshaping.

  5. Transistor: Transistors can be used for signal amplification, switching, and modulation.

  6. Potentiometer: A potentiometer (pot) can be used as a user-controllable knob to adjust the LFO frequency, waveform shape, or other parameters.

  7. LED: LEDs can be used for visual indication of the LFO signal, or for optical modulation.

  8. Photocell: Photocells can be used for optical modulation of the LFO signal, by varying the resistance based on the amount of light.

  9. Relay: Relays can be used for switching the LFO signal between different destinations, or for controlling other electrical components in the circuit.

  10. Transformer: Transformers can be used for signal isolation, impedance matching, or voltage regulation in the LFO circuit.

154> cool! could you show me an example of how to do each of these operations in the context of the design of an lfo?

Sure, here are some examples:

  • Capacitors: Capacitors are often used in filters to smooth the output signal of an LFO. For example, a low-pass filter can be used to remove higher frequency components from the output signal. The capacitance of the capacitor determines the cutoff frequency of the filter. In the circuit, capacitors are usually represented as two parallel lines:

    +------+
    |      |
    |      |
    +------+
    
  • Resistors: Resistors are used to limit current and voltage in the circuit. They can also be used in filters and feedback loops to modify the response of the circuit. In an LFO, resistors can be used to set the bias voltage of an op-amp or to adjust the gain of a filter. In the circuit, resistors are usually represented as a zigzag line:

    +------+
    |      |
    +-\/\/--+
    |      |
    +------+
    
  • Diodes: Diodes are used to allow current to flow in one direction only. In an LFO, diodes can be used to create a wave shaping circuit that modifies the shape of the output signal. For example, a diode clipper circuit can be used to create a square wave output. In the circuit, diodes are usually represented as an arrow pointing in the direction of current flow:

    +------+
    |      |
    |----->|
    |      |
    +------+
    
  • Transistors: Transistors are used to amplify or switch electronic signals. In an LFO, transistors can be used to control the flow of current in the circuit. For example, a transistor can be used to switch between different capacitors in a filter circuit to change the cutoff frequency. In the circuit, transistors are usually represented as a triangle with an arrow pointing in the direction of current flow:

    +------+
    |      |
    |  /|  |
    |/-+-\ |
    |/   \|
    +------+
    
  • Op-amps: Op-amps are used to amplify and process signals in the circuit. In an LFO, op-amps can be used to create filters, amplifiers, and other signal processing circuits. Op-amps are usually represented as a triangle with two inputs (+ and -) and one output:

           +------+
    in+---->|      |
           |      |
    in-+---->|      |----> out
           |      |
           +------+
    
  • Inductors: Inductors are used in filters to remove high frequency components from the output signal. They are also used in resonant circuits to create a specific frequency response. In an LFO, inductors are not commonly used due to their size and cost. In the circuit, inductors are usually represented as a coil:

         +-----+
         |     |
    in---|     |---out
         |     |
         +-----+
    

These are just a few examples of the many different components that can be used in an LFO. The specific components used will depend on the desired functionality of the circuit and the available resources.

155> interesting -- what are some others? i like the way you are doing it!

Here are a few more common operations involved in the design of an LFO:

  1. Voltage-Controlled Amplifiers (VCAs): These components allow for the amplitude of an LFO waveform to be controlled by a separate control voltage signal. VCAs are useful for modulating the intensity or depth of LFO modulation in a patch.

  2. Low-Pass Filters (LPFs): LPFs are used to smooth out an LFO waveform to reduce any high-frequency content or noise. This can be useful for creating a more subtle or natural-sounding modulation effect.

  3. Waveform Shaping Circuits: These circuits allow for the modification or shaping of LFO waveforms beyond the basic sine, square, triangle, and sawtooth shapes. Examples include waveshaping circuits that add harmonics or distortion to LFO waveforms, or circuits that can create complex or unpredictable waveforms.

  4. Envelope Followers: These circuits track the amplitude envelope of an incoming audio signal and generate a control voltage signal that can be used to modulate the LFO rate or other parameters. This can be useful for creating tempo-synced LFO effects that respond to the dynamics of an incoming signal.

  5. Trigger/Gate Generators: These circuits generate a trigger or gate signal based on incoming audio or control voltage signals. These signals can be used to synchronize the LFO to other parts of a modular patch, or to create rhythmic or patterned LFO effects.

156> how do we build a trigger gate generator?

A trigger gate generator can be built using a Schmitt trigger gate and a timing capacitor. The Schmitt trigger gate is a type of logic gate that has hysteresis, meaning that the threshold for switching from high to low or low to high is different. This can be useful for creating stable and precise triggers. The timing capacitor is charged or discharged by the output of the Schmitt trigger gate, and the rate at which it charges or discharges determines the frequency of the trigger signal.

Here's a simplified circuit diagram of a trigger gate generator:

                 +5V
                  |
                  R
                  |
                 Schmitt Trigger Gate
                  |
                  C
                  |
                 GND

The resistor (R) and capacitor (C) values are chosen to set the timing of the trigger pulses. When the Schmitt trigger gate output is high, the capacitor is charged through the resistor. When the output goes low, the capacitor discharges through the same resistor. The time it takes to charge and discharge the capacitor determines the frequency of the trigger pulses.

In order to use this circuit as a trigger gate generator for an LFO, we would need to feed the output of this circuit into the trigger input of the LFO circuit. The LFO would then generate a waveform with a frequency determined by the timing of the trigger pulses.

157> thanks! what about an envelope follower?

An envelope follower is a circuit that extracts the shape of an audio signal's amplitude envelope. The envelope can be used to control other circuits or processing elements, such as a voltage-controlled amplifier (VCA) or a filter.

Here is an example schematic for an envelope follower:

          R1
in  o-----/\/\/\---+
                   |
          R2       |       op-amp
   +----/\/\/\------+----o  output
   |               |
   |               |
   |               |
   C1             _|_
   |             ---   R3
   |              |    |
   +--------------+----o
                  |
                  |
                 GND

The input audio signal is applied to a voltage divider formed by resistors R1 and R2. The output of the voltage divider is then filtered by a capacitor C1, which removes the audio signal's high frequency components, leaving only the envelope. The resulting voltage is then amplified by an op-amp with a gain determined by R3.

The output of the envelope follower can be used to control other circuits, such as a VCA or a filter. For example, the output can be used to vary the gain of a VCA, causing the audio signal to be "ducked" or "pumped" in time with the amplitude of the input signal.

158> thanks! what about a waveform shaping circuit?

A waveform shaping circuit is a circuit that transforms an input waveform into a desired output waveform. The circuit can be designed to perform a variety of functions, such as distortion, filtering, frequency modulation, and waveshaping.

One common application of a waveform shaping circuit is to produce distortion effects, such as overdrive or fuzz, in electric guitar amplifiers. In this case, the circuit is often designed to clip or distort the input waveform in a specific way, depending on the desired sound.

Another application of waveform shaping circuits is in synthesizers and sound processing equipment. In these applications, the circuit can be designed to modify the amplitude, frequency, or phase characteristics of the input waveform, allowing for the creation of complex and unique sounds.

The specific design of a waveform shaping circuit will depend on the desired output waveform and the specific application. In general, the circuit will include one or more active or passive components, such as diodes, transistors, resistors, capacitors, or inductors, and may require careful tuning and calibration to achieve the desired output waveform.

159> could you show me a circuit diagram?

Certainly! Here's an example circuit diagram for a simple waveform shaping circuit:

        +------R1------+
        |               |
IN -----C1--R2--C2-----|----- OUT
        |               |
        +---------------+

In this circuit, IN is the input signal and OUT is the output signal. The circuit consists of two capacitors (C1 and C2) and two resistors (R1 and R2). When an input signal is applied to the circuit, the capacitors and resistors interact to modify the shape of the waveform.

The specific behavior of the circuit depends on the values chosen for the components. For example, if C1 is much larger than C2 and R1 is much larger than R2, then the circuit will act as a high-pass filter, attenuating low-frequency components of the input signal while allowing high-frequency components to pass through.

By adjusting the values of the components, it is possible to create a wide range of different waveform shaping circuits, including filters, equalizers, and distortion circuits.

160> cool! what about a voltage controlled amplifier?

A voltage controlled amplifier (VCA) is an electronic circuit that can vary its gain or attenuation of an input signal in response to a control voltage. It is a fundamental building block of many analog synthesizers, as it is used to shape the amplitude of an audio signal in response to a control signal, often an envelope or LFO.

The basic circuit for a VCA is a variable gain amplifier with a control voltage applied to the gain control input. The control voltage can come from an external source, or it can be generated by an envelope or other modulation source within the synthesizer.

One common circuit for a VCA is the "exponential VCA", which uses a diode-connected transistor to apply an exponential voltage to the gain control input. The output voltage of the circuit is proportional to the product of the input voltage and the exponential control voltage.

Here is a circuit diagram for a basic exponential VCA:

        +---------+
        |         |
  In----|         |
        |    Q1   |--------+--------> Out
  CV----|    Q2   |----+   |
        |    Q3   |----|--|
        +---------+    |  R
                        |  |
                        |  C
                        |  |
                        +--+

In this circuit, Q1 is a differential pair amplifier, Q2 is a voltage-controlled current source, and Q3 is a current-to-voltage converter. The control voltage is applied to the base of Q2, which controls the current flowing through R and C. This current is converted to a voltage by Q3, and this voltage is then applied to the base of Q1 to control the gain of the amplifier.

As the control voltage increases, the gain of the amplifier increases exponentially, which means that small changes in the control voltage can have a large effect on the output signal. This makes the exponential VCA a useful tool for shaping the amplitude of audio signals in response to modulation sources like envelopes and LFOs.

Dependencies

~11–20MB
~299K SLoC