John Riselvato

Here we’ll use an input file with a 1 minute duration with a 5 second introduction of silence. To remove that silence, it’s important to identify these two numbers because trimming requires tricky subtraction. Let’s look at the example below:

$ ffmpeg -ss 00:00:10 -i input.mp3 -t 00:01:00 -async 1 output.mp3   

-ss is used to seek to a part in the audio and -t is the duration of the playback. So 1 minute duration audio subtracted by 10 seconds seeking results in the first 10 seconds being removed.

A major setback with trimming is manually setting the audio duration value. Luckily, FFMPEG has a tool called ffprobe which can be used to give you exact duration:

$ ffprobe -i input.mp3 -show_entries format=duration -v quiet -of csv="p=0" -sexagesimal

Note: “How to Use FFMPEG in bash Language?” for ideas on how to use this output as a variable and dynamically set -t the advanced FFMPEG programming.

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Merging audio with FFMPEG is one of the most searched questions on the internet and couldn’t be any easier. The only difference is a txt file with a list of mp3 tracks must be made before the merging can begin. Below is an example list inside the file “file.txt”:

file 'track1.mp3'
file 'track2.mp3'
file 'track3.mp3'

Next, using the following FFMPEG one-liner, all 3 mp3 files can be merged into one long track:

  $ ffmpeg -f concat -i file.txt -c copy full.mp3   

concat or concatenate, is the process of combining media files. In this example, exactly the same codec and parameters are required for joining. 

In this example, ffmpeg lacks a specific media file as an input but a list from a external list (file.txt), this -f is used to “force” and input file.

NOTE: this will work for any audio or video format.

concat
	Indicates the argument that combines multiple media files
-f 
	Indicates the forces an input or output

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FLAC (Free Lossless Audio Codec) is another free audio codec and container that is known for it’s lossless audio quality. Oddly enough, FLAC is also maintained and developed by the xiph.org foundation. Other than FLACs inconvenience of player support, audiophiles consider FLAC to be one of the best formats for it’s Hi-Res audio.

Although this might be a superior audio quality for some, mp3 is just more convenient and thus converting a FLAC to mp3 is as follows:

 $ ffmpeg -i input.flac -ab 320k output.mp3  

In this example, the introduction of -ab or bitrate can be noted. The higher the bitrate the higher the quality but larger the file.

Note: this conversion will automatically keep metadata (FFMPEG 3.2 +)

Looking to convert an OGG to MP3? Check out this article.

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At times, you may have a video that you wish to extract the audio from. This is easily done by simply converting the video into an audio format. For example:

 $ ffmpeg -i input.mp4 -vn -acodec copy output.mp3 

A new argument -vn (no video) is used to ensure the encoded mp3 is only an audio file. FFMPEG is smart enough that this argument is optional in most cases. Here we also copy the input audio codec and use the same one in the output mp3.

If for any reason a conversion of a video to audio format results in a broken mp3, setting the mp3 codec to libmp3lame may fix the issue.

 $ ffmpeg -i input.mp4 -vn -acodec libmp3lame output.mp3 

FFMPEG does support just about any audio format you required including WAV, AIFF, FLAC, ALAC, ACC, WMA, etc. For a full list check out these other articles:

• What are all the formats FFMPEG supports?
• What are all the codecs FFMPEG supports?
• What is a Codec? explained

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The -map functionality is an advanced feature that allows the user to select specific input audio or video that is sent to the output. In addition, virtual streams can be created to apply complex manipulations which can be accessed with -map. 

Normally, a map isn’t required to set an output as the result audio and video is known:

 $ ffmpeg -i input.mp4 output.mp4 

In examples where multiple inputs are available, the use of -map is specific on what the output will include. In this example, the video (v) from input1.mp4 (index 0) is selected for output while the audio (a) from input2.mp4 (index 1) is selected for output:

 $ ffmpeg -i input1.mp4 -i input2.mp4 -map 0:v -map 1:a -shortest output.mp4 

In the case of two inputs with a filter, the syntax is a little different. In this example, the output will have the video input1.mp4 and the audio from input2.mp4 with a volume increase of 10dB for the output result:

 $ ffmpeg -i input1.mp4 -i input2.mp4 -filter_complex "[1:a]volume=10dB[new_audio]" -map 0:v -map [new_audio]:a -shortest output.mp4 

The volume filter is applied to [1:a], the audio in input2.mp4, and a new virtual stream, [new_audio], must be created for -map to set the new audio in the output. Accessing the audio from [new_audio] follows the same pattern with the [new_audio]:a.

If an input has multiple audio sources as seen in MKV files, the access to a specific audio channel syntax would be as seen below:

 $ ffmpeg -i input.mkv -map 0:v -map 0:a:1 output.mp4 

In addition to virtual streams being created, the virtual stream can also have a filter applied to it through chaining. For example, with the same concept as the last example except this time, the [new_audio] has a reverse filter applied:

 $ ffmpeg -i input1.mp4 -i input2.mp4 -filter_complex "[1:a]volume=50dB[new_audio]; [new_audio]areverse[reversed_audio]" -map 0:v -map [reversed_audio]:a -shortest output.mp4 

A new audio stream, [reversed_audio], is created and set for the output. The final results being video from input1.mp4 with audio from input2.mp4 with 10dB higher volume and reverse is now complete. 

The syntax of -map accesses to a stream’s audio and video is very similar to dictionaries in the Swift programming language. A dictionary has a key and value, [key:value], and in FFMPEG the key is the index/name while the value is either audio or video.

Note: If the output requires video and audio both must be mapped.

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