Hi my name is Diego F. from Quito Ecuador and this is my topic.

CATEGORIES OF EFFECTS DYNAMICS EFFECTS

Sounds, including music, can be barely audible, or loud enough to hurt your ears, or anywhere in between. When they want to talk about the loudness of a sound, scientists and engineers talk about amplitude. Musicians talk about dynamics. The amplitude of a sound is a particular number, usually measured in decibels, but dynamics are relative; an orchestra playing fortissimo sounds much louder than a single violin playing fortissimo. The exact interpretation of each dynamic marking in a piece of music depends on: comparison with other dynamics in that piece the typical dynamic range for that instrument or ensemble the abilities of the performer(s) the traditions of the musical genre being performed the acoustics of the performance space Traditionally, dynamic markings are based on Italian words, although there is nothing wrong with simply writing things like "quietly" or "louder" in the music. Forte means loud and piano means quiet. The instrument commonly called the "piano" by the way, was originally called a "pianoforte" because it could play dynamics, unlike earlier popular keyboard instruments like the harpsichord and spinet.

AMPLITUDE

Compressors
Compressors make loud sounds quieter and quiet sounds louder by decreasing or "compressing" the dynamic range of an audio signal.[63] A compressor is often used to stabilize volume and smooth a notes "attack" by dampening its onset and amplifying it is sustained. A compressor can also function as a limiter with extreme settings of its control.

Limiters
A limiter is a type of compressor designed for a specific purpose to limit the level of a signal to a certain threshold. Whereas a compressor will begin smoothly reducing the gain above the threshold, a limiter will almost completely prevent any additional gain above the threshold. A limiter is like a compressor set to a very high compression ratio (at least 10:1, more commonly 20:1 or more). The graph below shows a limiting ratio of infinity to one, i.e. there is no gain at all above at the threshold. Limiters are used as a safeguard against signal peaking (clipping). They prevent occasional signal peaks which would be too loud or distorted. Limiters are often used in conjunction with a compressor the compressor provides a smooth roll-off of higher levels and the limiter provides a final safety net against very strong peaks.

Expanders
An expander is closely related to a gate like a limiter is related to a compressor. An expander leaves louder information (that is above the threshold) untouched while simply turning down softer information when it goes below the threshold (instead of completely turning it off like a gate). The purpose of this is to increase the dynamic range of a signal (instead of decreasing it like a compressor). An expander is not really used too often, but can be useful in similar circumstances to the gate when cutting the signal off completely is a little extreme. I used an expander once in a voice over studio where they wanted to turn down the breaths, but not completely get rid of them.

Noise gates
Is used to control the volume of an audio signal. Gating is the use of a gate. Comparable to a compressor, which attenuate signals above a threshold, noise gates attenuate signals which register below the threshold. However, noise gates attenuate signals by a fixed amount, known as the range. In its most simple form, a noise gate allows a signal to pass through only when it is above a set threshold: the gate is 'open'. If the signal falls below the threshold no signal is allowed to pass (or the signal is substantially attenuated): the gate is 'closed'. A noise gate is used when the level of the 'signal' is above the level of the 'noise'. The threshold is set above the level of the 'noise' and so when there is no 'signal' the gate is closed. A noise gate does not remove noise from the signal. When the gate is open both the signal and the noise will pass

through. Gates typically feature 'attack', 'release', and 'hold' settings and may feature a 'look-ahead' function. They are commonly used in the recording studio and sound reinforcement. Rock musicians may also use small portable units to control unwanted noise from their amplification systems. Band-limited noise gates are also used to eliminate background noise from audio recordings by eliminating frequency bands that contain only static.

FILTER EFFECTS
he auto-filter (automatic filter) is a modulation effect that repeatedly sweeps a filter's cutoff frequency up and down to create tonal changes to a track. It's widely used in electronic to process drum and synch loops to create gradual changes to the mix. Modulated filters are a common feature in most synthesizers, and a standard effect available in most DAWs. How does it work? An auto-filter combines a filter and a low-frequency oscillator (LFO) used to modulate the filter's cutoff frequency. While most auto-filter plug-ins use a resonant low-pass filter, some plug-ins offer high-pass and band-pass filters as well. As the LFO rises and falls, it moves the filter's cutoff frequency up and down. With a low-pass filter, when the LFO rises and the cutoff frequency goes up, high frequencies in the input audio enter the mix. When the LFO falls and the cutoff frequency goes down, those high frequencies are muted and the effect's output becomes darker. Principal controls Auto-filter effects typically include controls to adjust the filter's cutoff frequency and resonance, and the speed and depth of an LFO to modulate the cutoff frequency. LFO speed settings are often slow for making gradual tonal changes, and LFO speeds are often synced to song tempo.

TIMBRE

High pass filter High pass filter is an electronic filter. It is also known as name HPF. These filters pass high frequency signals and blocks the low frequency signals. Generally they are used in devices. And attenuation of frequency depends on filter to filter. This model of filter is specially designed as a linear time invariant system. High pass filters are also known as bass cut filter or low cut filter and in some cases also known as rumble filters. Their names suggest their tasks. For example a high pass filter is used in audio system to transfer high frequencies to attain through cutting or filtering low frequencies. Even it is used with big or small speakers to low frequencies or eliminates bass. are the best option for frequency management, it is used to pass the high frequency signals and block lower frequency signals. These are also used for AC coupling at the inputs of audio amplifiers for protect application of direct current. In short these high pass filters are generally used in the circuits or in the devices where someone requires a high frequency signal. Low pass filter
The low pass filter only allows low frequency signals from 0Hz to its cut-off frequency, fc point to pass while blocking those any higher.

Band pass filter

The band pass filter allows signals falling within a certain frequency band setup between two points to pass through while blocking both the lower and higher frequencies either side of this frequency band.

Parametric EQ
The equalizer is an important piece of audio technology. As one of my Conservatorium tutors once said, when youre setting up a session, adding an EQ as the first insert is almost as essential as creating the tracks themselves. Studio audio isnt about capturing every frequency of every sound: its about creating a polished track that highlights the best of each instrument. Lets take a look at this basic yet widely misunderstood tool, the parametric EQ plug-in.

The Parametric EQ
Lets take a look at the plug-in itself. Im using the one that comes with Pro Tools LE, but you can use any parametric EQ in any DAW with these steps: There are several controls youll be using all the time, but three youll be using the most. Well cover those three in a second lets get the others out of the way first: In: the In button turns that particular EQ control on or off. If you equalize a frequency and then decide you dont want to keep the change, you can just turn off that EQ band until you need it, instead of having to zero out the settings. Shelf/Notch: the two buttons next to the EQ band name determine the shape of your EQ when it is at one end or the other of the frequency spectrum. Those in the middle are notch, meaning they affect a set range of frequencies, but these end bands can be shelved which means they are affected from the bottom of the frequency spectrum (for the LF) up to the set frequency, or the top of the spectrum for the HF.

Graphic EQ
A graphic Equalizer, more commonly known as an EQ is used to change the frequency response, or in other words the tone, of a sound, song, or instrument. It can be used to give more bass, less bass, more treble, etc. Set all the EQ bands on 0, or in the middle. This will make the audio come out of the speaker without any effects. Listen to your audio through the speaker to determine if it needs anything.2 Listen to your audio through the speaker to determine if it needs anything. Remember that the left side of the unit, usually starting around 20, is the low or bass side; the right, usually ending at around 16k, is the high or treble side. The mid is between 400 and 1.6k. Note that, once you have that down, you need to adjust the equalizer accordingly. Remember, when the equalizer is where you want it, you should turn the volume to desired level.

DELAY EFFECTS
This effect uses a RAM buffer to store the audio for a certain amount of time, after which it outputs this audio. The RAM buffer will not always be just large enough to store the amount of audio set by the current delay time, but will actually be considerably larger sometimes, because many delay effects allow for their delay parameter to automate.

PROPAGATION QUALITIES

Reverbs
Reverberation is crucial to our perception of the spaces we inhabit. Close your eyes, and your ears will tell you plenty about the space around you; the sound might be bright or dark, washed in echoes or dry or anywhere in between. When these clues are removed, as in a dead room or anechoic chamber, the results can be disconcerting. So, although we often record vocals or instruments dry to avoid capturing an undesirable room sound, we usually add ambience back in at the mixing stage using reverbs or

delays. But although it is easy to slap a reverb on an individual track and make it sound nice, getting reverb to work in the context of a mix is another story. In this article, we'll look at Steinberg's bundled reverb and delay plug-ins, and how best to exploit them within the DAW mixing environment.

Delays
Parameter Range: 0.000 to xxx.xxx seconds The upper boundary of this parameter (xxx.xxx) depends on the amount of RAM in the hardware unit, or the amount of RAM reserved by the software plug-in. For software plug-ins, the programmers or designers of the plug-in set this upper boundary. The delay software plug-in supertap by Waves for example comes in two flavors on Protools TDM systems. One allows for delays of up to two seconds, but another version accommodates delay times of up to six seconds. This is due to the availability of RAM on the DSP hardware of a TDM system. The native(->CPU) version of this plugin for Windows for example only allows for delays of up to four seconds in length, but there are delay plug-ins that allow for delays of several minutes.

Phasers
A phaser is an audio signal processing technique used to filter a signal by creating a series of peaks and troughs in the frequency spectrum. The position of the peaks and troughs is typically modulated so that they vary over time, creating a sweeping effect. For this purpose, phasers usually include a low-frequency oscillator. The electronic phasing effect is created by splitting an audio signal into two paths. One path treats the signal with an all-pass filter, which preserves the amplitude of the original signal and alters the phase. The amount of change in phase depends on the frequency. When signals from the two paths are mixed, the frequencies that are out of phase will cancel each other out, creating the phaser's characteristic notches. Changing the mix ratio changes the depth of the notches; the deepest notches occur when the mix ratio is 50%. The definition of phaser typically excludes such devices where the all-pass section is a delay line; such a device is called a flanger.[1] Using a delay line creates an unlimited series of equally spaced notches and peaks. It is possible to cascade a delay line with another type of all-pass filter as in,[2] this combines the unlimited number of notches from the flanger with the uneven spacing of the phaser

Flangers
Is an audio effect produced by mixing two identical signals together, with one signal delayed by a small and gradually changing period, usually smaller than 20 milliseconds. This produces a swept comb filter effect: peaks and notches are produced in the resultant frequency spectrum, related to each other in a linear harmonic series. Varying the time delay causes these to sweep up and down the frequency spectrum. A flanger is an effects unit dedicated to creating this sound effect. Part of the output signal is usually fed back to the input (a "re-circulating delay line"), producing a resonance effect which further enhances the intensity of the peaks and

troughs. The phase of the fed-back signal is sometimes inverted, producing another variation on the flanging sound.

Choruses
In music, a chorus effect (sometimes chorusing or chorused effect) occurs when individual sounds with roughly the same timbre and nearly (but never exactly) the same pitch converge and are perceived as one. While similar sounds coming from multiple sources can occur naturally (as in the case of a choir or string orchestra), it can also be simulated using an electronic effects unit or signal processing device.