Audacity Audio Editor Common Voice Recording Effects For Voiceover

Common effects used for editing sound in voice recording. They may be called other things in different brands of software, so the important thing is to understand the function.

To 'amplify' is to increase the loudness or volume of the selected region. To make a part of the recording softer or louder, select it and then use the menu "Effects / Amplify".

To 'normalize' is to adjust the volume so that the loudest peak is equal to (or a percentage of) the maximum signal that can be used in digital audio. Usually you normalize recordings to 100% as the last stage in production to make it the loudest possible without distortion.

Compressor (Dynamic Range Compressor)
A Dynamic Range Compressor limits the volume levels of a voice recording so that it stays within a certain loudness range.
An example of where it is used is in TV broadcasting, where it ensures that the volume levels of ads are perceived as being louder than the television program itself (without any change in the actual broadcast volume).
It also has a use for recording audio from one medium to another, where the two mediums are not capable of handling the same range of volume levels (e.g. A CD can handle a much greater range than a cassette tape).
To apply this effect, select a region then use the menu "Effects / Compressor" to bring up the Dynamic Range Compressor settings.
The Threshold setting works by detecting when the sound recording volume exceeds a defined decibel level. It then gradually attenuates the sound to bring it down below the dB level, and does it in such a way that the listener will not be aware the attenuation is occurring.
The Limit setting defines at what maximum decibel level the sound recording will be allowed to rise up to. So if, for example, the Limit was set to 0dB, then you will never hear the volume level of the recording get louder than 0dB. The Limit setting has similarities to the Threshold setting, but the main difference is that the Threshold does allow sounds to go above the defined decibel level (for a short time), whereas the Limit does not.
You will find that the minimum Limit volume you can set is the same as the maximum Threshold value. This basically means that, in any situation, the sound will start to attenuate at the threshold level, but will never be heard louder than the limit.
The Ratio setting limits the amount the volume level of the recording increases at any one time. If, for example, you wanted the volume levels of a recording to only increase by at most 1/4 of the amount they would normally increase, then this would correspond to a Ratio of 4:1. So if the recording volume level increased by 8dB, then you would only hear a 2dB volume increase.
The buttons for General Voice Level and TV / Radio Advertisement are preset settings appropriate for these types of recordings.

An equalizer changes the frequency response of a signal so it has different tonal qualities.

To assist you with shaping the Equalizer graph in the way you want, there is typically a preset list that displays the most common sorts of filters used in the Equalizer graph. You can choose any preset filter from the list and then manipulate the filter to achieve the effect you desire. The list of filters to choose from and how you can shape them are explained below. Note that all fields where a frequency value is entered can have a maximum value of the highest frequency the equalizer is programmed to work with, typically 20,000 HZ

Graphic Equalizer
The Graphic Equalizer uses discrete sliders to set the gain or attenuation of a signal at a particular frequency. You can select how many sliders you would like to manipulate by entering a value between 3 and 20 in the box at the top of the display. When you change the number of sliders you would like to utilize, the frequencies are automatically allocated to best span the audible frequency range from 20Hz to 20kHz. Selecting presets allows you to easily configure common filters such as low pass or high pass.

Parametric Equalizer
The Parametric Equalizer is similar to the Graphic Equalizer, but with more control. Here you can adjust the frequency and bandwidth of the individual sliders by left clicking on the frequency or Q values below each slider. Frequency must be set between 20Hz and 20,000 Hz. The Q parameter must be set between 0.05 and 20. A higher Q causes the gain or attenuation peak at the frequency to be much sharper, and therefore less likely to impact adjacent frequency content, while a lower Q applies the modification more smoothly across the frequency spectrum. making the peak shafrper enables you to remove a hum for example, while leaving the rest of the audio content as undisturbed as possible

The 'envelope' is the change in volume of the select region over time. This can be used to make fine adjustments to the volume over time or even more crude changes like fade in or fade out.

Stereo Pan
The stereo pan effect allows you to change how loud the sound is that comes out the left or right speaker. For example if you had a stereo recording with all the sound coming out of only one speaker, you could use the pan effect to "center" the sound yourself. You can also make a centered sound change move one from speaker to the other as the sound file plays. Select the region you want to change the pan for and choose Effects -> Stereo Pan. Click on a point and move it upwards for an increase in volume on the left speaker, or move it downwards for an increase in volume on the right speaker. Please note the stereo pan effect only works on stereo files. If your file is not stereo you must first convert it to stereo.
This effect reverses the selection in the same way playing a record or tape backwards would.

An echo is a repeat of the sound after a short time (usually 400 - 1000ms). It sounds a bit like the person is in a large stadium or is shouting between two mountains.
To add echo select the region and use the menu "Effects / Echo", then specify the duration and amplitude of the echo. The duration is the length of time after which the sound repeats - usually this is between 400 and 1000ms. The amplitude can be between 1 - 99% (99 being a very loud echo).

Reverb is many small reflections of the sound that come after a set time. It usually occurs when someone is speaking in a room, hall etc. More reverb is called wet, no reverb is called dry.
The Reverb effect allows you to adjust the reverb level and time. The reverb level is the amplitude - 99 is very wet, 0 is dry. The time can be between 100 and 800ms - 200ms sounds like a small room or 800ms a large hall.
If you add too much reverb it can sound like the person is in a pipe or in the bathroom.

The phaser sound effect is created by mixing a slightly delayed signal with the original. You can set the delay in ms (typical default 5ms) and the wet dry gain in percent. 100% is wet. 0% is off/dry.

A Flanger sound effect is similar to the phaser except that the delay is slowly modulated over time. You specify the starting delay time (typical default 5ms), the frequency of modulation in times per second (typical default 0.5Hz which is 2 seconds) the depth of modulation (typical default 50%) and the wet dry gain (100% for wet, 0% for dry).

The tremolo sound effect is similar to the vibrato effect, except that the amplitude pulsates rather than the pitch. The higher the Frequency (Hz) set, the more often the pulsation will be heard, and the higher the Depth (%), the deeper the fluctuation in volume.

The vibrato sound effect is a pulsating of the pitch at a depth and frequency specified by the user. The higher the Frequency (Hz) set, the more often the pulses will be heard, and the higher the Depth (semitones), the wider the fluctuation in pitch will be.

The doppler effect simulates the sound of a passing vehicle, which has a high pitch while approaching, shifting to a low pitch when traveling away from the listener. Specify the Velocity (in km/h) of the passing source; a higher velocity will result in a higher starting pitch and lower ending pitch. Adjust the Listener Horizontal and Vertical Positions to indicate the listener's horizontal and/or vertical position to the passing source; play around with the values to achieve different combinations of pitch.

As the name suggests, the effect modulates a specified frequency band within the sample, which results in the characteristic "Wah wah" sound. The effect is a bandpass filter with its center frequency (not to be confused with the center frequency parameter, below) alternating between a min frequency and max frequency (specified by the center frequency and depth parameters) and from max frequency to min frequency. The frequency of alternating direction is represented as a triangular wave with a frequency specified by the wah frequency parameter.
Resonance: also known as Q or emphasis, this parameter controls the resonant peak of the bandpass filter. This value determines the sharpness of the wah-wah effect. Higher values produce more resonant/peaky tones.
Depth: this parameter determines the frequency range swept by the bandpass filter. Its range is specified as a percentage of the range (0 to center frequency). If the value of the percentage of the range (0, center frequency) is specified as X, the min and max frequencies are (center frequency - X) and (center frequency + X).
Center Frequency: This parameter is the center frequency of the bandpass filter sweep, and is used to determine the min and max frequencies as mentioned above.
Wah Frequency: This is the frequency of alternating the direction of the sweep, or the frequency of the wah-wah sound. It is the frequency of the triangular wave described above.

The chorus sound effect is used to make one voice or one instrument sound like 3 voices or instruments by playing the original with variably delayed and slightly pitch changed copies of the original.
Note: Chorus is a very useful way to make a mono source sound more stereo. You should convert your file to stereo first before using Chorus.

AM Radio
This simulates an AM Radio. Iit accurately simulates a 'good' AM radio. To make it worse, apply the effect twice. For a really bad sound, paste mix some soft white noise (use the Tone Generator tool) to simulate bad reception.

This simulates the audio down a telephone line. It simulates a 'good' telephone line. To make it worse apply the effect twice and paste mix soft white noise.

Fade (Fade In, Fade Out, Fade Out and Trim)
To fade in or out the recording use the menu "Effects / Fade / Fade In" or "Effects / Fade / Fade Out".
The Fade Out and Trim option is a combined function which fades out over the selection then marks the end of the selection as the end of the recording. This is frequently used at the end of music tracks.

Simple Speed and Pitch Change
This plays the recording faster or slower which in turn increases or decreases the pitch too. This function is useful to correct slow or fast tapes.

Speed Change
Normal speed changes (i.e. "Simple Speed and Pitch Change" above) changes the pitch in proportion to the speed. If you want to change the speed but keep the pitch the same use this function. Speed can change the duration of the audio. The time duration (in seconds) can also be adjusted using this effect.

Pitch Change
This changes the pitch of the recording without changing the speed (i.e. the converse of the above). Change of semitones can also be adjusted using this effect

Pitch Speed Profile
This allows you to specify how much to change pitch, speed, or pitch and speed at any point in the file, using a graph.

Auto Gain
Normal recordings can have the volume of the recording too high in parts and too soft in parts. Automatic Gain Control reduces the too loud parts and increases the too soft parts. This is sometimes a better alternative to normalization.

High Pass (High-Pass Filter)
A high-pass filter (sometimes called a low cut filter) removes all low frequencies below a specified Hz. This is useful if you want to make your recording sound 'clearer' or less 'muddy'. It is very usual to use a high-pass filter of about 300Hz on all voice recordings to improve intelligibility.

Low-Pass Filter
A low-pass filter removes all high frequencies above a specified Hz. This is useful if you want to make your recording sound 'clearer'. It is very usual to use a low-pass filter of about 1600Hz on all voice recordings to improve intelligibility.

DC Offset Correction
Often when you record audio using bad electronics the recording has a constant 'DC' level throughout the file. Because the ear cannot hear this you will not notice it until you attempt to edit in other audio when you can hear horrible clicks. If you think this is the problem you can run DC Offset Correction over the entire recording before you begin to edit. Another (and possibly better) way to deal with this problem is to run a high pass filter (say at 50Hz) over the recording.

Auto Click/Pop Removal
This tool allows you to apply a repair of a single click/pop artifact. To use it properly, you must zoom right in to the artifact and select a small region around it. Then select Tools menu -> Auto Click/Pop Removal. The repair will be performed straight away.

Reduce Vocals
If you want to reduce the vocals from a music track you can use this effect. WavePad will attempt to identify the voice in the left-to-right spectrum of a stereo recording and remove it. The recording must be stereo (from an original stereo source like a CD - simply converting a file to stereo will not work). It will also remove any instruments near the voice in the stereo spectrum.
Note: it is impossible to remove the vocals perfectly without the original mix track. You will notice some instruments might be removed too and some vocal remain. The effect will also not work on some files which have previously encoded in a highly compressed form like mp3 (because this remove some stereo depth).

Noise Reduction
Audacity has a noise removal tool that operates by taking a sample of the noise, the by reopening the tool the noise can be removed. Use the default settings to start with.
Also called: "Spectral Subtraction" method

  1. "Multiband Noise Gating" method - fast but inaccurate. Sometimes using both (Spectral Subtraction always must be first) then Multiband Noise Gating works very well.

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