Amplify...

This effect increases or decreases the volume of a track or set of tracks. When you open the dialog, Audacity automatically calculates the maximum amount you could amplify the selected audio without causing clipping (from being too loud).

Bass Boost...

This is a safe, smooth filter which can amplify the lower frequencies while leaving most of the other frequencies alone. It is most effective if you don't try to boost too much; 12 dB is usually just right.

Echo...

A simple delay line.

This effect repeats the audio you have selected again and again, softer each time. There is a fixed time delay between each repeat.

First select the audio you want to apply the effect to. You may want to first add silence to the end of your track(s) so that the echo has plenty of time to die out. When you select "Echo..." from the Effect menu, Audacity will ask you for two numbers.

The first number is the amount of delay between the echos, in seconds. The second value is the decay factor, which is a number between 0 and 1. A decay factor of 0 means no echo, and a decay factor of 1 means that each echo is just as loud as the original. A value of 0.5 means that its amplitude is cut in half each time, so it dies out slowly. Smaller values will make it die out even more quickly.

The Echo effect is very simple and is not intended to be used in place of a Reverb effect, which simulates the sound of a room, concert hall, stage, or other natural environment. Audacity for MacOS and Windows comes with FreeVerb, a free VST Reverb plug-in. See FreeVerb, below, for more information.

Note that if you set the decay value to 1.0, you can use Echo to create loops that repeat as long as you want any never change volume.

Fade In

Applies a linear fade-in to the selected audio. For a logarithmic fade, use the envelope tool.

Fade Out

Applies a linear fade-out to the selected audio. For a logarithmic fade, use the envelope tool.

FFT Filter...

This effect is fully functional but the dialog box is unfortunately still under construction. You can still use it, but there are no axes to tell you which frequencies are which, or how much gain you're applying.

This is the most general type of filter. If you're careful, you can use it to highlight exactly the frequencies you want. However, doing an FFT filter is more likely to result in artifacts, especially if the filter you draw is not smooth.

Invert

This effect flips the audio samples upside-down. This normally does not affect the sound of the audio at all. It is occasionally useful, for example when the left and right channels of a song both contain equal amounts of vocals, but unequal amounts of background instruments. By inverting one of the channels and not the other, the vocals will cancel each other out, leaving just the instrumentals. Obviously this only works if the exact same vocal signal is present in both of the channels to begin with.

Noise Removal...

This effect is ideal for removing constant background noise such as fans, tape noise, or hums. It will not work very well for removing talking or music in the background.

Removing noise is a two-step process. In the first step, you select a portion of your sound which contains all noise and no signal, in other words, select the part that's silent except for the noise. Then choose Noise Removal... from the Effect menu and click Get Profile. Audacity learns from this selection what the noise sounds like, so it knows what to filter out later.

Then, select all of the audio where you want the noise removed from and choose Noise Removal... again. This time, click the "Remove Noise" button. It may take a few seconds or longer depending on how much you selected.

If too much or not enough noise was removed, you can Undo (from the Edit menu) and try Noise Removal... again with a different noise removal level. You don't have to get a new noise profile again if you think the first one was fine.

Removing noise usually results in some distortion. This is normal and there's virtually nothing you can do about it. When there's only a little bit of noise, and the signal (i.e. the voice or the music or whatever) is much louder than the noise, this effect works well and there's very little audible distortion. But when the noise is very loud, when the noise is variable, or when the signal is not much louder than the noise, then the result is often too distorted.

Future versions of Audacity may include improved versions of this effect.

Phaser...

The name "Phaser" comes from "Phase Shifter", because it works by combining phase-shifted signals with the original signal. The movement of the phase-shifted signals is controlled using a Low Frequency Oscillator (LFO).

Reverse

This effect reverses the selected audio temporally; after the effect the end of the audio will be heard first and the beginning last. Some people reverse small portions of audio to make inappropriate language unintelligible, while others believe you can hear subliminal messages if you listen to speech backwards. You can also create interesting sound effects by recording natural events and reversing the audio.

Wahwah...

Just like that guitar sound so popular in the 1970's.

This effect uses a moving bandpass filter to create its sound. A low frequency oscillator (LFO) is used to control the movement of the filter throughout the frequency spectrum.

The WahWah effect automatically adjusts the phase of the left and right channels when given a stereo recording, so that the effect seems to travel across the speakers.

Audacity for Mac OS and Windows comes with a VST plug-in called Freeverb, which is in version 2 (hence "Freeverb2"). This effect implements a versatile and high-quality reverb effect.

Getting a good reverb sound depends a lot on the source audio and can take a lot of experimentation. One good strategy is to select a small portion of audio (a few seconds) and try to add reverb. Listen to it, then Undo and try it again with different settings. Keep doing this until you've found the settings you like the sound of best, and then Undo one last time, Select All, then apply the effect to your entire recording.

There are a lot of parameters to Freeverb2: Room size, Damping, Predelay, Lowpass, Highpass, Wet level, and Dry level.

Let's start with the wet and dry levels. Freeverb works by taking your audio signal and modifying it to create the reverberated sound, i.e. the sound you hear echoing off the walls of the room. This is called the "wet" part of the effect.

This is mixed together with the original sound, which is called the "dry" part of the effect, to produce the combination of direct (unreverberated) and indirect (reverberated) sound, which is what you would naturally hear. If you set the dry level to -infinity and the wet level to 0 dB, the result is something like standing outside of a concert hall: you can only hear the reverberated sound.

If you set the dry level to 0 and the wet level to -infinity, it's like standing right in front of the singer in a tiny room - all you can hear is the singer, and no reverb. A good place to start is to set both the dry level and wet level to 0 dB. However, you may want to experiment with lowering the dry level while you are experimenting with the sound of the reverb.

The room size parameter is self-explanatory. The smallest room size setting creates a quick, bright reverb, while the largest setting creates a long, drawn-out, and dark reverb. The damping parameter controls how the sound bounces off the walls - i.e. if it is mostly reflected or absorbed.

The predelay controls the delay between the dry signal (unreverberated) and the wet signal (reverberated) - usually there is some predelay because of the time it takes sound waves to travel from the sound source to the nearest wall, and to the microphone. Larger predelays are suitable for creating an effect of a larger room.

Finally, the lowpass and highpass filters can be used to make the reverberated sound lower or higher. Increasing the lowpass filter filters out the high frequencies, and similarly increasing the highpass filter filters out the low frequencies.