Digital reverberation is a continually relevant and widely discussed topic in the realms of computer music and Digital Signal Processing, as well as electroacoustic music in general. Its applications and studies have involved both commercial and academic sectors. Consequently, over time, a complex history has developed, characterized by numerous ramifications and implications, leading to a proliferation of various methods and implementation topologies. In this study, we will delve into the subject in detail, examining the main existing implementations.
Reverberation is the persistence of sound after it has been produced. It is an acoustic phenomenon related to the reflection of sound waves by an obstacle placed in front of the sound source. Assumptions that determine the perception of a reverberation phenomenon:
Given these assumptions, in an open space, reverberation can be discussed when the obstacle is less than 17 meters from the sound source. Indeed, up to this distance, the path of the sound wave from the source to the obstacle and back is less than 34 meters, and therefore the sound takes less than 100 milliseconds to return to the starting point, blending into the listener's ear with the original sound. If the obstacle is more than 17 meters away from the source, then the delay of the reflected sound compared to the direct sound is more than 100 milliseconds, and the two sounds are therefore distinct. In this case, it is called an echo.
The factors that influence the duration of reverberation are multiple. The most influential ones are:
Room size
Materials
For these reasons related to materials, a small room like a bathroom has longer reverberation times than a large wooden room.
The best way to listen to the reverberation of a reverberant space is to produce an impulsive sound; like a clap of hands or a snap of fingers.
Music has made extensive use of reverberation for thousands of years. Archaeologists believe that reverberation produced by caves was used in ancient ceremonies. Many cathedrals in Europe have reverberations lasting more than 10 seconds, and the choral music of certain eras worked particularly well by exploiting the reverberation inside these cathedrals. In fact, the reverberation of individual notes overlaps on subsequent notes, transforming a monophonic melody into a polyphonic sound.
A standard room has 6 surfaces:
A sound, when produced, bounces off the surfaces and is subsequently heard, producing what are called:
Each of these will produce another 6 echoes: 6 echoes, each bouncing off the 6 surfaces, will produce 36 echoes; these are called:
producing a total of 42 echoes in a very short period of time, and so on... Of all these echoes, none is perceived individually, but rather their ensemble and dispersion over time are perceived. Reverberation is thus composed of thousands of echoes of the original sound that persist and decay over time.
Reverberation is artificial when it is not present in the room where the recording is taking place but is instead added later.
Tape echo A particular magnetic tape recorder/player is used, which constantly moves a tape loop inside a mechanism with a fixed recording head and a mobile playback head. The signal recorded by the first head is read by the second and mixed with the original, generating the effect. These devices are bulky and heavy. Like in any tape recording, there is background noise similar to hiss, significantly higher than that produced with digital technologies.
Spring reverb The signal is passed, through a transducer, through a metal spiral (the spring). At the other end of the spring, a transducer equivalent to the first one reintroduces the signal into the amplification circuit, mixing it with the original. The signal taken from the second transducer is slightly delayed compared to the one applied to the first, creating the reverberation effect in the listener's ear.
Chamber reverb Following the spring reverb model, in a box acoustically isolated from the outside, a curved tube is inserted to create the longest possible path. At one end of the tube is placed a small loudspeaker, while at the other end there is a microphone. The sound emitted by the loudspeaker will take some time to travel the entire tube and reach the microphone, thus generating the necessary delay. The signal taken from the microphone will be fed back into the mixing console, mixed with the original.
Plate reverb Similar to spring reverb, but with a large metal plate instead of the spring. It has two transducers attached to its surface and works in a similar way, although its quality is significantly higher.
They are produced by a computer or dedicated DSP integrated circuits. There are integrated circuits on the market that include A/D and D/A converters, memories, and timing circuits. An acoustic signal is transduced and converted into numbers that enter memories. In fact, the bytes are "scrolled" from one bank to the next until the last one is reached. The digital signal taken from the last memory is then reconverted into analog and mixed with the original signal, obtaining the reverberation effect; the farther the read point from the write point, the longer the echo time will be. The size of these read and write memories is called delay line, and it is expressed in samples. The large capacity of RAM memories allows achieving delays of several seconds and therefore smoothly transition from reverb to echo. The strategy used afterwards is to feed back the output of the delay line by adding it to the input, thus creating a feedback circuit. All this process is done because modern computers are very powerful, but they are not yet powerful enough to generate all the reflections heard in a large room, one by one. Rather, the goal of creating digital reverbs is to implement models and strategies to replicate the impression of the reverberation of a room. The replication process has generated in the history of digital reverbs true and proper typical sounds different from each other which can be implemented and preferred by musicians for aesthetic reasons.
Experiments and algorithms of digital reverb models in the FAUST language (GRAME)
Delay lines in Faust are divided into the following categories: mem - indicates a single sample delay @ - indicates a number (e.g., 44100) of variable delay samples x'- x indicates any input and: ' a sample delay, ''(2), etc. rdtable - indicates a read-only table rwtable - indicates a read and write table.
Through delay lines, we can create a Dirac impulse, which represents our minimum unit, namely the single sample by putting a number 1 and subtracting the same value from it but doing it at a delayed sample.
We will illustrate 3 main methods:
Writing the code line with internal recursion:
in this way the tilde ~ operator sends the signal output to itself, to the first available input creating a feedback circuit. One way to force the operator to point to a certain point in the code, is to put parentheses (), in this way ~ will point to the input before the parenthesis.
A second method consists of using with{} .
You can define a function in which are passed the various arguments of the function that control the parameters of the code, and say that that function is equal to exit from the with with ~ _ example:
function_with(argument1, argument2) = outwith ~
with{
section1 = _ argument1;
section2 = argument1 argument2;
out_with = section2;
};
where out_with ~ _ returns to itself.Moreover, with in Faust allows declaring variables that are not pointed to from outside the code but only from the belonging function; in this case the function to which with belongs is "function_with".
A third method is to use the letrec environment.
with this method we can write a signal recursively, similar to how recurrence equations are written.
Function for Conversion from Milliseconds to Samples: we input the time in milliseconds, and the function gives us the value in samples.
For example, if we have a sampling frequency of 48,000 samples per second, it means that 1000ms (1 second) is represented by 48,000 parts, and therefore a single unit of time like 1 ms. Corresponds digitally to 48 samples.
For this reason, we divide the sampling frequency by 1000ms, resulting in a total number of samples that corresponds to 1 ms. in the digital world at a certain sampling frequency.
And then we multiply the result of this operation by the total number of milliseconds we want to obtain as a representation in samples. If we multiply *10. For example, we will get 480 samples at a sampling frequency of 48,000 samples per second.
Function for Conversion from Samples to Milliseconds: we input a total number of samples, of which we need to know the overall duration in milliseconds based on our sampling frequency.
We know that a sampling frequency corresponds to a set of values that express together the duration of 1 second (1000 ms).
It means, for example, that at a sampling frequency of 48,000 samples per second, 1000 milliseconds are represented by 48,000 parts. So if we divide our 1000ms. / into the 48,000 parts which are the samples of our system, we would get the duration in milliseconds of a single sample at that sampling frequency, in this case therefore: 1000 / 48,000 = 0.02ms. And so the duration in milliseconds of a single sample at 48,000 samples per second, is 0.02 milliseconds. If we multiply the obtained number * a total number of samples, we will get the time in milliseconds of those samples for that sampling frequency used.
Obviously, as can be deduced from the considerations, as the sampling frequency increases, the temporal duration of a single sample decreases, and thus a greater definition.
In the digital domain, the feedback of a delay line, when applied, costs by default one sample delay. Feedback = 1 Sample
At the moment I decide therefore to put inside the feedback a number of delay samples, we can take for example 10 samples in our delay line, it means that, The direct signal will come out for delay samples at:
input in the delay signal --> output from the delay 10samp
1st Feedback: output from the delay at 10samp + 1 feedback = input in the delay 11samp --> output from the delay 21samp
2nd Feedback: output from the delay at 21samp + 1 feedback = input in the delay 22samp --> output from the delay 32samp
3rd Feedback: output from the delay at 32samp + 1 feedback = input in the delay 33samp --> output from the delay 43samp
and so on...
we can therefore notice immediately that we will not have the correct delay value required inside the same, because of the sample delay that occurs at the moment when I decide to create a feedback circuit. if we use the method of subtracting one sample from the delay line, we will have this result:
input in the delay signal --> -1, output from the delay 9samp
1st Feedback: output from the delay at 9samp + 1 feedback = input in the delay 10samp --> -1, output from the delay 19samp
2nd Feedback: output from the delay at 19samp + 1 feedback = input in the delay 20samp --> -1, output from the delay 29samp
3rd Feedback: output from the delay at 29samp + 1 feedback = input in the delay 30samp --> -1, output from the delay 39samp
and so on...
we can therefore notice that with this method, compared to the previous one we will have as input to the delay line always the number of delay samples required. But we notice that from the first output of the delayed signal subtracting -1 we have one sample delay less than we would like. To realign everything, we just need to add one sample delay to the overall output of the circuit, thus having from the first output:
input in the delay signal --> -1, output from the delay 9samp +1 = 10out
1st Feedback: output from the delay at 9samp + 1 feedback = input in the delay 10samp --> -1, output from the delay 19samp +1 = 20out
and so on...
Let's proceed with an implementation:
The term "T60" in the context of digital reverberation refers to the reverberation time. The reverberation time is a measure of the duration for which sound persists in a space after the sound source has stopped. It indicates how quickly the sound energy decreases over time.
The T60 value represents the time it takes for the sound level to decrease by 60 decibels (dB) compared to its initial value. In other words, it is the time taken for the sound energy to decay by 60 dB. A long T60 indicates prolonged reverberation, while a short T60 indicates shorter reverberation.
The formula below uses the relationship between the T60 decay time and the number of filter samples to calculate the amplification gain necessary. The result of the calculation is a linear value ranging from 0 to 1, representing the amplification to be applied to the filter feedback.
Insert the following arguments into the function:
The value in samples of the filter you are using for the delay.
The decay value in T60 (decay time of 60 dB in seconds)
= GET as output from the function, the value to be passed as amplification to the filter feedback to achieve the desired T60 decay time
represents the input signal, ( denotes the signal) it is then split into two parallel paths <: one delayed by one sample ' (' denotes one sample delay) and one without delay , (, denotes transition to the second path) they are then summed into a single signal :> _ ; the delayed signal has a feedforward amplitude control feedforward there is a general amplitude control outgain on the output function onezeroout
+~ is the summation, and the feedback of the arguments inside parentheses () represents the input signal, ( denotes the signal) delayed by one sample _ (automatically in the feedback) which enters : into the gain control of the feedback 1-feedback the same feedback controls the input amplification of the signal not injected into the feedback there is a general amplitude control outgain on the output function onezeroout
TPT version of the One-Pole Filter by Vadim Zavalishin reference: (by Will Pirkle) http://www.willpirkle.com/Downloads/AN-4VirtualAnalogFilters.2.0.pdf
represents the input signal, ( denotes the signal) it is then split into two parallel paths <: one delayed by @(delaysamples) samples (thus value to be passed externally) and one without delay , (, denotes transition to the second path) they are then summed into a single signal :> ;
In the feedback, one sample of delay is already present by default, hence delaysamples-1.
the delayed signal has a feedforward amplitude control * feedforward
there is a general amplitude control * outgain on the output function onezeroout
+~ is the summation, and the feedback of the arguments inside parentheses () represents the input signal, ( denotes the signal) delayed by @(delaysamples) samples (thus value to be passed externally) which enters : into the gain control of the feedback * feedback
In the feedback, one sample of delay is already present by default, hence delaysamples-1.
there is a general amplitude control * outgain on the output function combfeedbout
similar to the comb filter, but within the feedback, following the feedback enters the signal : into the onepole. The onepole is a lowpass where the cutoff frequency can be controlled between 0. and 1. In the feedback, one sample of delay is already present by default, hence delaysamples-1.
from the sum (+ transitions : to a cable _ and a split <: then @delay and gain, in feedback ~ to the initial sum. filtergain controls the amplitude of the two gain states, which in the filter are the same value but positive and negative, one side -filtergain and one side +filtergain. In the feedback, one sample of delay is already present by default, hence delaysamples-1. To maintain the delay threshold of the value delaysamples, a mem delay (of the subtracted sample) is added at the end
D. Rochesso, “Reverberation,” DAFX - Digital Audio Effects, Udo Zölzer, 2002.
Introduction to Digital Filters: With Audio Applications. books by Julius O. Smith III. Links to the series by Smith:
TOM ERBE - UC SAN DIEGO - REVERB TOPOLOGIES AND DESIGN http://tre.ucsd.edu/wordpress/wp-content/uploads/2018/10/reverbtopo.pdf
ARTIFICIAL REVERBERATION: su DSPRELATED Artificial Reverberation | Physical Audio Signal Processing
Corey Kereliuk - Building a Reverb Plugin in Faust Keith Barr’s reverb architecture Building a Reverb Plugin in Faust
Spin Semiconductor DSP Basics Spin Semiconductor - DSP Basics Spin Semiconductor Audio Effects Spin Semiconductor - Effects
freeverb3vst - Reverb Algorithms Tips http://freeverb3vst.osdn.jp/tips/reverb.shtml
History of allpass loop / "ring" reverbs History of allpass loop / "ring" reverbs? - Spin Semiconductor
Musical Applications of Microprocessors (The Hayden microcomputer series) http://sites.music.columbia.edu/cmc/courses/g6610/fall2016/week8/Musical_Applications_of_Microprocessors-Charmberlin.pdf
Acustica_Riverbero - Alfredo Ardia Appunti: acustica_Riverbero
Algorithmic Reverbs: The Moorer Design Algorithmic Reverbs: The Moorer Design | flyingSand
Dattorro Convex Optimization of a Reverberator Dattorro Convex Optimization of a Reverberator - Wikimization
primes under 10.000 https://www.matematika.it/public/allegati/34/Numeri_primi_minori_di_10000_1_3.pdf