Table of Contents

1.How would I take a section of my input file and save it to separate file?
2.When I open NEW on the central box what is your recommendation for frequency?
3.What is audio codec?
4.What is sampling frequency?
5.How to put reverb onto some parts of a recording, rather than having to put it on the whole recording?

How would I take a section of my input file and save it to separate file?

To save a part of a song as a separate file:

Step 1 Open your input file in AVS Audio Editor
Step 2 Select the part of your input file you want to save
Step 3 Choose Copy from Edit menu
Step 4 Press Ctrl+N to create a new file
Step 5 Go to Edit menu and select Paste
Step 6 Save the new file.

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When I open NEW on the central box what is your recommendation for frequency?

You can set it as 44100 Hz as set by default - your output file will be CD-quality. Actually the higher frequency - the higher the quality and the larger size of your output file. The sampling frequency is essentially the number of times the sound event is quantized within a given time period. Sampling frequencies are specified in KiloHertz (KHz), a term meaning samples per second. For example, "CD-quality" sound requires 16-bit words sampled at 44.1 KHz. Essentially this means 44,100 16-bit words (705,600 bits) are used to digitally describe each second of sound on a compact disc. The highest pitch possible is 22.05 KHz (approximately the top of human hearing range), which is half of 44.1 KHz.

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What is audio codec?

A video or audio Codec (COmpression/DECompression) is a software component allowing to encode data to be stored on a media (CD, DVD, etc...) and/ to decode it to be visualized or heard. Codecs can be implemented in software, hardware, or a combination of both.

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What is sampling frequency?

Sampling frequency also impacts fidelity. The sampling frequency is essentially the number of times the sound event is quantized within a given time period. Sampling frequencies are specified in KiloHertz (KHz), a term meaning samples per second. The key is understanding how sampling frequency affects fidelity is the Nyquist sampling theorem. Basically, when applied to audio signals the Nyquist theorem states that the highest possible pitch in the sound is one-half that of the sampling frequency.
For example, "CD-quality" sound requires 16-bit words sampled at 44.1 KHz. Essentially this means 44,100 16-bit words (705,600 bits) are used to digitally describe each second of sound on a compact disc. The highest pitch possible is 22.05 KHz (approximately the top of human hearing range), which is half of 44.1 KHz.

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How to put reverb onto some parts of a recording, rather than having to put it on the whole recording?

You can apply any effects to a part of your input file. How to apply:

Step 1 Open input file in AVS Audio Editor.
Step 2 Find the part of your file you want to put reverb onto
Step 3 Select (highlight) the part of your file you want to put reverb onto
Step 4 Go to Transform menu ->Delay Effects->Reverb ->...

In this case the effect will be applied at the selected part of your file.

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