WAV files are great while you are recording or editing: they are uncompressed, every editor opens them, and nothing is lost between saves. The trouble starts when you need to share them. A few minutes of stereo WAV can easily be larger than an email attachment limit, and many podcast hosts, chat apps and older car stereos would rather get an MP3. Converting is simple, but a few choices along the way decide whether the result sounds right and plays everywhere.
1. Pick a bitrate based on the content, not a habit
MP3 is lossy, so the bitrate is the main quality dial.
- Speech (voice notes, interviews, lectures): 96 to 128 kbps is usually plenty, and mono is fine if it was recorded with one microphone.
- Music or mixed content: 192 kbps is a sensible default; go up to 256 or 320 kbps if the file is a final master someone will listen to on good headphones.
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Variable bitrate (VBR): with LAME,
-q:a 2gives transparent results for most listeners and keeps quiet passages small.
Going above 320 kbps isn't possible in MP3, and encoding a low-quality source at 320 kbps doesn't bring back detail that was never there.
2. Keep the sample rate unless you have a reason to change it
Most WAV recordings are 44.1 kHz or 48 kHz. MP3 handles both, so leave it alone. Resampling 48 kHz to 44.1 kHz is only worth it when a target device or platform specifically asks for it. Files at 96 kHz from studio sessions are the exception: downsample them before encoding, because MP3 can't store that rate anyway.
3. Watch out for 24-bit and 32-bit float sources
Exports from DAWs are often 24-bit or 32-bit float. The encoder converts them internally, but if the original was mixed hot and peaks above 0 dBFS in float, the MP3 can clip. Normalising to about -1 dBFS before encoding avoids that, and it also leaves headroom for the small overshoots MP3 decoding can introduce.
4. Don't forget the metadata
WAV files rarely carry useful tags, while MP3 players lean heavily on ID3. Add at least a title and artist (or a show name for podcasts) after converting, otherwise the file shows up as "Unknown" in every library.
5. A quick command-line version
If FFmpeg is installed, this covers most cases:
ffmpeg -i input.wav -codec:a libmp3lame -b:a 192k output.mp3
Swap -b:a 192k for -q:a 2 if you prefer VBR.
6. When you'd rather not install anything
Not everyone on a team has FFmpeg set up, and uploading private recordings to a random converter isn't always acceptable. We built Convertola's WAV to MP3 converter for that situation (disclosure: it's our project). It runs the conversion in the browser, so the file stays on your device, and it lets you choose the bitrate before downloading.
Wrapping up
For most people the safe recipe is: keep the original sample rate, pick 128 kbps for speech or 192 kbps and up for music, normalise anything that peaks too close to zero, and add basic tags. Keep the WAV as your master copy and treat the MP3 as the version you share.
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