In this post, you will learn the best practices on digital audio editing. This is written for beginners in digital music. The practices outlined in this tutorial can be applied to any audio editing software either in Windows, Mac OS X or Linux. The practices below are also applicable in professional music production environment.
Tip#1: Always work on the highest audio resolution as possible
A common beginner mistake is using a lossy file type such as MP3 when editing audio files. While this is possible, this is not the best practice because MP3 file is already a compressed version.
Compressed version does not contain complete information of the audio that you are going to manipulate in the audio editing process. The best practice is to use the highest audio resolution as the source audio file for editing. This is usually the uncompressed audio file format such as .wav or .aiff file.
Case Example:
1.) When you are converting a WAV file to MP3. You will get better sounding results if you are converting from an uncompressed file type such as WAV. An even much better quality if you are converting from 24-bits WAV file than the 16-bit audio version (also known as the CD audio version).
If you do not have the 24-bit WAV file version, you can still use the 16-bit version and still works out fine for most cases.
This practice is definitely helpful if you are distributing MP3 files. Supposing you both have a 16-bit WAV and 128kbps MP3 version of the song; you should not use the 128kbps MP3 version to create a 320kbps MP3.
The best practice is to always use the highest resolution available and that is the 16-bit WAV version. This method would produce better sounding MP3 audio files.
2.) This rule is not only helpful in creating MP3’s but editing audio waveforms as well. Supposing you would want to edit the waveform of Song X; you should be using the uncompressed version (16-bit WAV file will do but 24-bit WAV version is highly recommended).
3.) If you really don’t have any original WAV version of your song; you can use the MP3 version and import it to your software. Take note that your software can’t natively process MP3 files with effects or plug-in; so the software will convert that to an uncompressed format (.wav).
Bear in mind this conversion process won’t restore the quality lost in the original MP3 conversion. It is simply a method that allows you to edit compressed audio files in your software.
Tip#2: Always save your work as uncompressed digital audio format
After completing all audio editing tasks, you should always be saving the edited audio as uncompressed audio format (WAV or AIFF); ideally the same format as the source audio file (if the source is uncompressed). Never save it to lossy/compressed format such as MP3. Refer to the table below:
Not lossy format
The table recommends that you should not be saving the edited file to a lossly/compressed format. For example if you are editing 24-bits WAV file, you should also be saving it as 24-bits WAV. Or if you are editing MP3 file because you don’t have any high resolution version, then save it either as 16-bits or 24-bits (recommended) WAV file after editing. This will preserve the quality after editing.
In Mac OS X, the equivalent uncompressed audio file format is AIFF. So if you are editing an MP3 file and would like to create another MP3 version; the following would be the recommended conversion flow:
MP3 source — > 16-bits WAV version (saving it) — > MP3 encoder — > Edited MP3
There are different types of MP3 encoders. The most recommended MP3 encoder is LAME; you can install it as plug-in at your audio editing software. Read the following example tutorial on how to add LAME MP3 encoder:
How to Add LAME MP3 encoder in Adobe Audition or Cool Edit Pro
Reaper DAW Tutorial
LAME for MP3 is an open source/free software.
In addition, when it says “sample rate should be the same with the source audio”; implies that you should not be converting the sample rate when saving the editing file. For example if the source audio is 16-bits/44.1kHz WAV. The sample rate in this case is 44.1 KHz. After saving the file, the sample rate should still be the same.
This practice preserves the original quality of your source audio as sample rate conversion can introduce audible artifacts after processing.
Tip#3: Never clipped beyond 0dBFS
When you are editing a digital audio waveform; there are times when the result would boost or amplify the original signal. This is when you are using effects such as EQ, amplification, compression, etc. When applying effects; make sure your digital audio waveforms would never clip beyond 0dBFS. This will result to audible distortion. Clipping is easy to spot because this will turn the level meters to red (inside the white box).
Clipping on Reaper
The most recommended approach is to leave at most -3dBFS headroom. See the waveform screenshot below:
Headroom specs
As you can see, the red line is the maximum possible digital audio level and that is 0dBFS. Beyond this level, it is called “clipping”. Make sure that when you are doing audio editing, you should be careful that there are no clipped waveforms. At a start, you can set the maximum peak to -3dBFS to make sure you still have enough headroom/allowance for future editing. Find out more about digital clipping in the following tutorials:
Recording Clipping Prevention Techniques
Tip#4: Always backup your original source audio before editing
Audio editing can be destructive. This means that saving the file would overwrite the original file with and there is no way to bring it back. Beginners often commit this mistake only to realize that they want to undo all the changes which is not possible after saving the file.
As the best practice, backup all audio files before doing the editing process. When you are saving files and you accidentally overwrite the file, you can repeat the editing process by using the backup files.
Tip#5: If possible, always use non-destructive editing method
Non-destructive editing method preserves the original audio file during the editing process. The effects are only remembered and applied on the software and not directly rendered to the original audio file saved in your hard drive.
Not all audio editing software offers this feature. If you want a non-destructive audio editing environment, you can use REAPER. Below is an example of controlling the audio file volume in different sections of the edited waveform by using the volume envelope/automation. This feature is also available in other audio editing software:
Reaper volume non destructive editing
Volume automation is used to control the loudness in different sections of the waveform. For example in the above screenshot, the audio is edited such that there is silence at the beginning of the audio file. Envelope is a very flexible and easy audio editing tool.
This type of editing is considered safe because it won’t alter the original source audio. When you are finally implementing the changes, REAPER would ask you for a new file name so that original file won’t be overwritten in the process. For more information on this topic, read this post on editing audio files: non-destructive vs. destructive editing.
Tip#6: If possible, always use minimal effects processing
Effects processing can further add errors and artifacts particularly if the source audio are lossy formats (MP3). You should keep the processing to a minimum if this is the case.
There are instances when you need to do a lot of processing on the digital audio. In this case, make sure you are working with a high resolution WAV file (24-bits) as they are an accurate/complete representation of original audio.
It is why 24-bits audio is an industry standard in professional music production. Engineers working in the studio are often applying a lot of effects to improve digital audio quality. Since the source audio is uncompressed and high resolution; quality would be preserved and digital errors are kept to a minimum during the editing process.
Content last updated on July 23, 2012