Change Audio Sample Rate with FFmpeg on Linux

This guide provides a quick, practical overview of how to change the audio sample rate of a media file using FFmpeg on a Linux system. You will learn the core command-line syntax for basic conversions, how to modify sample rates while preserving multiple audio tracks, and how to batch-process an entire directory of files.

Understanding the Audio Resampling Command

FFmpeg uses the -ar flag to define the target audio sample rate (measured in Hz). When you change the sample rate, FFmpeg automatically handles the resampling process using its internal resampler.

The standard syntax for a basic conversion is:

ffmpeg -i input.mp3 -ar 44100 output.mp3

In this command:

• -i input.mp3: Specifies the path to your source file.
• -ar 44100: Sets the new audio sample rate to 44,100 Hz (CD quality). Other common values include 48000 (DAT/DVD quality) and 96000 (High-Resolution audio).
• output.mp3: The name and format of the newly generated file.

Advanced Resampling Scenarios

Depending on your project requirements, you may need to control the audio codec or handle multiple audio streams within a video container.

Resampling Audio Within a Video File

If you want to change the audio sample rate of a video file (like an MP4 or MKV) without re-encoding the video stream, you must explicitly tell FFmpeg to copy the video data. This saves a massive amount of processing time and prevents quality loss.

ffmpeg -i input.mp4 -c:v copy -ar 48000 output.mp4

The -c:v copy flag instructs FFmpeg to copy the video stream exactly as it is, while the -ar 48000 flag alters the audio stream to 48 kHz.

Explicitly Defining the Audio Codec

By default, FFmpeg selects a native encoder based on your output file extension. If you want to force a specific audio codec during the resampling process, use the -c:a flag:

ffmpeg -i input.wav -c:a libmp3lame -ar 44100 output.mp3

Batch Processing Multiple Files in Linux

If you have a large folder of audio files that all need to be converted to the same sample rate, you can leverage the Linux Bash shell to automate the task.

Run the following loop directly in your terminal to process all .wav files in a directory and save them as 44.1 kHz .mp3 files:

for file in *.wav; do
    ffmpeg -i "$file" -ar 44100 "${file%.wav}_resampled.mp3"
done

This script loops through every file ending in .wav, applies the -ar 44100 flag, and appends _resampled.mp3 to the original filename to avoid overwriting your source data.