transcoding audio with ffmpeg

src: FFmpeg - Ultimate Guide | IMG.LY Blog

In this chapter, we’ll be going to see how to transcode into audio with FFmpeg!

The general formula is:

ffmpeg -i {input audio or video file with audio} [output options] output_audio.ext

Choosing a format

FFmpeg is quite smart, and by the extension, it can determine which codec to use. If you specify “audio.wav” or “audio.mp3” for example, FFmpeg will use the appropriate codec to do the encoding.

It is perfectly guessing most of the time. But if you want to specify the format manually, then the “-f” flag is your friend.

For this, you might want to consult the list of formats:

ffmpeg -formats

So, these three commands will do exactly the same, but the last two requires the -f flag.

# Output codec is determined from the extension
ffmpeg -i bbb_audio.wav bbb_audio.mp3
 
# No extension in the filename
ffmpeg -i bbb_audio.wav -f mp3 bbb_audio
 
# Piped output therefore no filename, so no extension to use for guessing
ffmpeg -i bbb_audio.wav -f mp3 pipe:1 > bbb_audio

Setting the bitrate

In most cases. you want to specify the target bitrate you expect from your codec to output. If you are unsure what bitrate is, please read this article’s audio bitrate section.

To specify the audio bitrate, use the “-b:a” option with a corresponding value, e.g.:

• -b:a 320k: For the mp3 codec this is considered high quality.
• -b:a 128k: Lower quality.
• -b:a 64k: Low quality.

For example:

ffmpeg -i bbb_audio.wav -b:a 320k bbb_audio_320k.mp3

Setting the sample rate

You may want to specify the sample rate to ensure quality or low output file size. Half the sample rate could mean half the output file size. If you are unsure what the sample rate is, please read the “audio sample rate” section of this article.

To specify the audio sample rate, use the “-ar” option with a corresponding value, e.g.:

• -ar 48000: For high quality.
• -ar 44100: For CD quality (still high).
• -ar 22500: A bit of a compromise, not recommended for music, but for speech, it might be enough.
• -ar 8000: Low quality, e.g. if you only want “understandable” speech.

For example:

ffmpeg -i bbb_audio.wav -ar 44100 bbb_audio_44100khz.mp3

Setting the channel count

Setting the channel count can be useful, for example, if you have a stereo recording of a single person’s speech. In that case, you might be content with just a mono output half the size of the original recording.

If you are unsure what an audio channel is, please read the “audio channels” section of this article.

To specify the channel count use the  “-ac” option with a corresponding value, e.g.:

• -ac 1: For mono
• -ac 2: For stereo
• -ac 6: For 5.1

For example:

ffmpeg -i bbb_audio.wav -ac 1 bbb_audio_mono.mp3

Complete command line for converting audio with FFmpeg

This is how you produce a high-quality output:

# Convert wav to mp3
ffmpeg -i bbb_audio.wav -ac 2 -ar 44100 -b:a 320k bbb_audio_hqfull.mp3
 
# Convert wav to m4a (aac)
ffmpeg -i bbb_audio.wav -ac 2 -ar 44100 -b:a 320k bbb_audio_hqfull.m4a
 
# Convert wav to ogg (vorbis)
ffmpeg -i bbb_audio.wav -ac 2 -ar 44100 -b:a 320k bbb_audio_hqfull.ogg

Check out this documentation about good quality audio transcoding too!.

Lossless formats

If you want to convert audio into a lossless format, here are a few choices for you:

# Convert to flac (Free Lossless Audio Codec)
ffmpeg -i bbb_audio.wav -compression_level 12 bbb_audio_lossless_12.flac # Best compression, slowest
ffmpeg -i bbb_audio.wav -compression_level 5 bbb_audio_lossless_5.flac   # Default
ffmpeg -i bbb_audio.wav -compression_level 0 bbb_audio_lossless_0.flac   # Least compression, fastest
 
 
# Convert to wav
cp bbb_audio.wav bbb_audio_lossless.wav # Just kidding:)
 
# Convert to wav 
ffmpeg -i any_audio.ext bbb_audio_lossless.wav

It’s good if you know that flac results in a smaller file than WAV, as WAV doesn’t actually compress by default:

117M bbb_audio.wav
52M  bbb_audio_lossless_0.flac
45M  bbb_audio_lossless_5.flac
43M  bbb_audio_lossless_12.flac

WAV is generally thought of as a lossless format, but keep in mind that the WAV container can contain lossy content too, but by default FFmpeg uses the pcm_s16le format, which is the 16 bit PCM, that could be understood as lossless.

Learn more here and here.