29 releases
0.2.12-alpha.0 | Apr 7, 2023 |
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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 |
#91 in #synthesizer
200 downloads per month
Used in 25 crates
(13 directly)
145KB
2.5K
SLoC
Surge-lag
Description
The surge-lag
crate is a Rust implementation of
a lag processor. The purpose of the lag processor
is to smoothly transition between different
parameter values in a signal processing chain.
The Lag
struct represents the lag processor, and
it has methods to instantiate it, set the rate of
change, set the starting value, and process input
values. The VLag
struct is a variant of Lag
that uses VDouble
values.
The Lag
processor can be used to create smooth
transitions between different parameter values in
a synthesizer, such as a filter cutoff frequency
or oscillator frequency. It can also be used to
create gradual fades in and out of signals.
The implementation is based on the following equation:
y[n] = y[n-1] + rate * (x[n] - y[n-1])
where x
is the input signal, y
is the output
signal, and rate
is the rate of change.
Tokens
-
Lag
: The struct representing the lag processor. -
VLag
: The struct representing the lag processor usingVDouble
values. -
default
: A method to create a defaultLag
instance. -
instantize
: A method to initialize aLag
instance with a specific starting value. -
set_rate
: A method to set the rate of change for aLag
instance. -
start_value
: A method to set the starting value for aLag
instance. -
process
: A method to process an input value through aLag
instance. -
VDouble
: A struct representing a double value using SIMD operations. -
init_x87
: A method to initialize aVLag
instance with a specific starting value using x87 instructions. -
test_vlag_init_x87
: A test to verify the initialization of aVLag
instance using x87 instructions. -
test_vlag_instantize
: A test to verify the initialization of aVLag
instance with a specific starting value. -
test_vlag_process
: A test to verify the processing of an input value through aVLag
instance. -
test_vlag_start_value
: A test to verify the setting of the starting value for aVLag
instance. -
smoke
: A test to verify the basic functionality of aLag
instance. -
test_default
: A test to verify the creation of a defaultLag
instance. -
test_instantize
: A test to verify the initialization of aLag
instance with a specific starting value. -
test_process
: A test to verify the processing of an input value through aLag
instance. -
test_set_rate
: A test to verify the setting of the rate of change for aLag
instance. -
test_start_value
: A test to verify the setting of the starting value for aLag
instance.
107> so this is like a sort of linear interpolator, with a variable rate. why do we use sines and cosines for its implementation?
The use of sines and cosines in the implementation
of the VLag
struct is related to the concept of
phase. In this implementation, the phase of a sine
and cosine wave is used to determine the
fractional part of the delay time. The delay time
is then calculated using linear interpolation
between two consecutive samples based on the
fractional part of the delay time.
The use of sines and cosines in this way is actually a common technique in digital signal processing, known as "phase accumulation". By accumulating phase values over time, we can create a variable time delay that is continuously changing in a smooth and continuous way. This can be useful in a variety of audio processing applications, such as reverberation, chorus, and flanging effects.
108> could you describe phase accumulation in more detail? could you show me how to derive the delay time computation?
Sure! Phase accumulation is a technique used in many digital signal processing applications to generate or process signals with specific frequencies. The basic idea is to keep track of a phase variable that increases over time, and then use this phase variable to compute the amplitude of the output signal at each sample.
In the context of the surge-lag crate, phase accumulation is used to implement a variable delay line. The phase variable is directly proportional to the delay time, so by controlling the rate of phase accumulation, we can control the delay time.
To derive the delay time computation, let's start with the definition of the phase variable phi:
phi(n) = 2 * pi * f * n * T
where n
is the sample index, T
is the sample
period, and f
is the frequency we want to
generate. To get the delay time, we need to solve
for n
in terms of the delay time d
:
n = d / T
Substituting this into the expression for
phi(n)
, we get:
phi(d) = 2 * pi * f * d
This is the formula for the phase at delay time
d
, in radians. To compute the amplitude of the
output signal, we need to take the sine or cosine
of this phase value:
y = A * sin(phi(d))
or
y = A * cos(phi(d))
where A
is the amplitude of the output
signal. By changing the rate of phase
accumulation, we can change the delay time, and
thus the frequency of the output signal.
I hope that helps! Let me know if you have any more questions.
109> thanks -- how do these techniques apply to creating digital reverberation, chorus, and flanging effects? are there other signal processors which can benefit from these functionalities?
The techniques used in the surge-lag
crate,
particularly phase accumulation and linear
interpolation, are commonly used in digital signal
processing algorithms for creating various types
of time-based effects, including digital
reverberation, chorus, and flanging.
In digital reverberation, for example, multiple
reflections of a sound wave are created to
simulate the effect of sound bouncing off walls
and other surfaces in a physical space. Each
reflection is delayed by a certain amount, and the
delayed reflections are then summed together with
the original sound to create a reverberant
sound. The delays are typically implemented using
some form of a delay line, such as the Lag
structure in the surge-lag
crate.
In chorus and flanging effects, multiple, slightly detuned copies of the original sound are created and mixed together with the original signal to create a thick, swirling sound. The detuning is typically implemented by introducing a small delay and slightly altering the phase of the delayed copy. Again, this can be implemented using a delay line with a variable delay time.
Other signal processors that can benefit from these functionalities include time-based effects such as echo, delay, and phaser, as well as oscillators and filters that require phase coherence.
110> let's describe some of these in more detail: echo, delay, and phaser, as well as oscillators and filters that require phase coherence
Sure, I'd be happy to describe these in more detail:
-
Echo: An echo effect creates a delayed copy of the input signal that is repeated at regular intervals. The delayed copy is typically attenuated to simulate the effect of sound reflecting off surfaces and returning to the listener. This effect can be created using a delay line, which is a buffer that stores a portion of the input signal and outputs it after a specified delay time. The delayed signal can then be combined with the original signal to create the echo effect.
-
Delay: A delay effect is similar to an echo effect, but with a shorter delay time and fewer repeats. Delay effects are commonly used to create a sense of space and depth in a mix. Delay can be implemented using a delay line, just like an echo effect, but with a shorter delay time and fewer repeats.
-
Phaser: A phaser is a type of signal processor that creates a sweeping filter effect by combining two delayed copies of the input signal. The first copy is passed through a variable all-pass filter, which shifts the phase of certain frequency components, while the second copy is passed through a fixed all-pass filter with a different phase shift. The two signals are then combined to create a series of peaks and dips in the frequency response of the processed signal, which creates the characteristic phaser effect.
-
Oscillators: Oscillators are electronic circuits that generate periodic waveforms, such as sine waves, square waves, and sawtooth waves. These waveforms can be used to create a variety of sounds, from simple tones to complex timbres. Phase coherence is important when using multiple oscillators to create polyphonic sounds, such as chords or multiple voices. If the phase of the oscillators is not synchronized, it can create unwanted phase cancellations and frequency interference.
-
Filters: Filters are signal processing circuits that selectively attenuate or boost certain frequency components of a signal. Filters are commonly used in audio processing to shape the frequency response of a signal, either to correct for unwanted resonances or to create specific timbres. Filters that require phase coherence, such as comb filters or phasers, must maintain the phase relationship between the input and output signals to avoid unwanted phase cancellations or frequency interference. This is often achieved by using all-pass filters, which shift the phase of the signal without affecting its magnitude, or by using phase-locked loops, which synchronize the phase of the input and output signals.
111> thanks -- what is the mathematical idea of phase coherence? how do we know when a system will exhibit problems with phase coherence?
Phase coherence refers to the property of a signal processing system where the relative phases of its output signals are preserved with respect to its input signals. In other words, if a system is phase coherent, then the phase relationships between different components of the input signal are preserved in the output signal.
A system can exhibit problems with phase coherence when there are differences in phase shifts among different frequency components of a signal. This can occur in systems that use different processing paths for different frequency bands or in systems that use feedback loops, among others. If the phase shifts are not properly compensated for, the output signal can become distorted and lose its coherence with respect to the input signal. This can result in a variety of undesirable effects, including phase cancellation, comb filtering, and other artifacts that can degrade the quality of the signal.
112> thanks -- could you provide 10 examples of projects which may benefit from using this crate?
Sure, here are 10 examples of projects that may
benefit from using surge-lag
:
-
Audio plugins and digital audio workstations (DAWs) that require time-based effects like echo, delay, phaser, or flanger.
-
Synthesizer software that requires phase coherence for its oscillators and filters.
-
Music production software that requires precise timing of events or synchronization with external sources like MIDI or SMPTE timecode.
-
Game audio engines that require accurate timing of sound effects and music.
-
Virtual reality and augmented reality software that requires accurate spatialization of sound sources.
-
Interactive installations and multimedia performances that involve real-time audio processing and manipulation.
-
Music visualization software that uses audio analysis to generate visual effects.
-
Audio hardware devices that require precise timing and synchronization with other devices.
-
Research projects in digital signal processing, acoustics, and psychoacoustics that require precise time-based processing.
-
Audio-related machine learning projects that require accurate timing and synchronization of audio signals.
113> thanks -- could you provide 10 others?
Sure! Here are 10 more examples of projects that could benefit from using the surge-lag crate:
-
Audio analysis software that requires phase-aligned signals for accurate measurements
-
Audio restoration software that needs to preserve the phase coherence of the original recordings
-
Digital signal processing libraries that implement time-based effects like echo, delay, and chorus
-
Music production software that uses digital signal processing for virtual instruments and effects
-
Audio synthesis software that generates sounds with complex timbres and dynamic textures
-
Machine learning algorithms that process audio data for speech recognition or music analysis
-
Virtual reality and augmented reality applications that require 3D spatialization of sound
-
Signal processing hardware for audio applications, such as mixers, amplifiers, and preamps
-
Educational tools for teaching digital signal processing and audio engineering
-
Digital art installations that use sound as an integral part of the user experience.
114> okay! some of these overlap, but that is ok
No problem, happy to help!
115> what is the difference between Lag and VLag?
From what I can see in the tokens you provided,
both Lag
and VLag
are structs that represent
a delay line used for creating time-based effects
like echoes and reverbs.
However, it seems that VLag
is a more
specialized version of Lag
, designed
specifically for linear interpolation of the delay
line, while Lag
may use different interpolation
methods or not perform any interpolation at all.
Without seeing the code or documentation for these
specific crates, I can't say for sure what the
exact differences are, but based on the tokens you
provided, it appears that VLag
is a more
specialized version of Lag
designed specifically
for linear interpolation of the delay line.
116> /file surge-lag/src/vlag.rs
The VLag
struct is similar to Lag
, but it uses
double-precision floating-point numbers (f64
)
instead of single-precision floating-point numbers
(f32
). It also supports SSE/AVX vectorization,
meaning that the filter's two double-precision
floating-point values (v
and target_v
) are
aligned on a 16-byte boundary to facilitate
efficient processing by modern CPUs.
The VLag
struct represents a variable lag
filter, which is used to smooth out a stream of
data over time, by updating the current value
(v
) based on the target value (target_v
) and
a fixed lag coefficient (VLAG_D_LP
), which
determines how quickly the filter responds to
changes in the target value. The lag coefficient
and the inverse of the lag coefficient
(VLAG_D_LPINV
) are pre-defined constants.
The VLag
struct has four public methods
(new_value
, instantize
, start_value
, and
process
) and several pre-defined constants
(VLAG_MIN_BW
, VLAG_D_LP
, and VLAG_D_LPINV
)
that are used to control the behavior of the
filter.
The VLag
struct is useful for smoothing out
signals in digital audio processing, especially
for signals with low frequency content. It can be
used in a variety of applications, such as
implementing a tremolo effect on a guitar signal,
or smoothing out the output of a sensor in
a robotics application.
117> /file surge-lag/src/lag.rs
Lag
is a struct that provides a low-pass filter
that smooths out a stream of data over time. It
achieves this by updating the current value based
on the target value and a fixed low-pass filter
coefficient, which determines how quickly the
filter responds to changes in the target value.
The struct is generic over a type parameter T
,
which must implement the Num
, From<f32>
,
Clone
, and AddAssign
traits. This allows Lag
to be used with a variety of numeric types.
The Lag
struct has several methods, including
new
, set_rate
, new_value
, start_value
,
instantize
, process
, and
get_target_value
. These methods allow the user
to initialize and manipulate the filter, including
setting the low-pass filter coefficient, setting
the target value to be smoothed towards, updating
the current value based on the target value and
filter coefficient, and retrieving the current
target value.
In addition, Lag
has a constant boolean value
FIRST_RUN_CHECKS
that determines whether to run
first-time checks, and a struct field first_run
that tracks whether the Lag
struct is being used
for the first time.
Dependencies
~11–20MB
~296K SLoC