Audio Convert

Any format → OGG Vorbis

ffmpeg -i input.mp3 -acodec libvorbis -q:a 5 output.ogg

Any format → OPUS (high quality, low bitrate)

ffmpeg -i input.wav -acodec libopus -b:a 96k output.opus

Any format → M4A (AAC)

ffmpeg -i input.mp3 -acodec aac -ab 192k output.m4a

Adjust Audio Properties

Change sample rate (e.g., to 16 kHz for speech recognition)

ffmpeg -i input.wav -ar 16000 output.wav

Convert stereo to mono

ffmpeg -i input.wav -ac 1 output.wav

Change bit rate

ffmpeg -i input.wav -ab 128k output.mp3

Combine sample rate + channels + format in one command

ffmpeg -i input.mp3 -ar 16000 -ac 1 -acodec pcm_s16le output.wav

Batch Conversion

Convert all MP3 files in a directory to WAV

for f in *.mp3; do ffmpeg -i "$f" "${f%.mp3}.wav"; done

Check File Info Before Converting

ffprobe -v quiet -show_streams -select_streams a:0 input.mp3

Common Codec Reference

Notes

• ffmpeg overwrites output files by default; add -n to skip existing files or -y to always overwrite.
• To strip metadata from the output, add -map_metadata -1.
• On resource-constrained PicoClaw hardware, prefer lightweight formats (WAV, MP3) over computationally expensive ones (FLAC encoding is fast; OPUS encoding may be slow on very low-end boards).