Why only WAV as input?

WAV contains raw PCM audio data that can be read directly by the browser’s `AudioContext.decodeAudioData`. The format supports 16-bit, 24-bit and 32-bit float without quality loss. Other formats like AAC, OGG, FLAC or M4A would require bundling separate decoders. To convert between compressed formats, use the original source or a desktop tool like ffmpeg.

What is the difference between mono and stereo?

Mono has one audio channel, stereo has two (left and right). For pure speech, stereo gives no benefit and needlessly doubles file size. Mono is the standard for radio news, single-voice podcasts and telephony. Stereo matters when there is spatial information such as instruments panned wide, vocals with panning, or nature field recordings. "Keep original" preserves the same channel count as the WAV file.

Is MP3 still a good format in 2026?

MP3 is over 30 years old but still the most compatible audio format on earth: every browser, media player, car, DJ setup and streaming platform supports it. Patents expired in 2017, so the format is now free. Newer formats like AAC (slightly better quality at low bitrates), Opus (best for voice and streaming) and FLAC (lossless) offer technically superior performance but lack MP3’s universal reach. For maximum compatibility, pick MP3; for better quality at the same size, pick AAC or Opus.

Are my files sent to a server?

No. Both WAV decoding (browser AudioContext) and MP3 encoding (LAME/lamejs in a Web Worker) run one hundred per cent locally on your device. Once the page has loaded you can turn off the internet and the tool still works. No file bytes leave your computer during upload or download. You can confirm this in the developer tools by inspecting the Network tab while converting.

/audio-converter

WAV to MP3 Converter - Free Local Audio Converter

Convert WAV audio files to MP3 format locally in your browser with adjustable bitrate and channel settings.

Drop a .wav file here, or click to browse

Only .wav files are supported. Processing is 100% local inside your browser.

Conversion Settings

Bitrate (quality)

Channels

How the conversion works

This tool makes it possible to convert uncompressed WAV files into compressed MP3 files quickly and easily. The entire process happens locally on your own device:

Client-side decoding: The browser's built-in audio API decodes the WAV file's raw PCM audio data. This supports all common WAV formats (including 16-bit, 24-bit and 32-bit float).
Background Encoding with Web Worker: A background thread (Web Worker) encodes the audio data into MP3 using the popular open-source LAME library (lamejs). Because this runs in the background, your browser tab won
100% Privacy: No data is uploaded to any external server. Your audio files remain on your device at all times, and the conversion requires no internet access once the page has loaded.

Which bitrate should I choose?

The bitrate determines the audio quality and file size of the MP3 file:

128 kbps: Excellent for voice, podcasts, or when you need the smallest possible file size.
192 kbps: A great standard for music, offering high quality while keeping files relatively compact.
256 kbps / 320 kbps: For high-fidelity music and audiophiles who want minimal quality loss compared to the uncompressed source file.

About this tool

The WAV to MP3 converter turns uncompressed WAV audio files into compressed MP3 files entirely in the browser. Files never leave your device. The conversion uses the open-source LAME library (lamejs) in a background thread (Web Worker), so the interface stays responsive even for long tracks. Pick a bitrate from 128 to 320 kbps and convert to mono, stereo or keep the original. Great for voice recordings, spoken word, podcast source material and music to be compressed for web or mobile.

How to use it

Drop a WAV file on the drop zone or click to browse your device.
Choose bitrate (kbps) and channels (mono/stereo or keep original).
Press "Convert to MP3" and watch the progress bar while the file is processed in the background.
Download the finished MP3 file. No data was sent over the network.

Examples

Podcast episode: 60 min recording

InputWAV stereo 44.1 kHz 16-bit, 60 min = 635 MB, bitrate 128 kbps mono

OutputMP3 approx. 55 MB (12x smaller). Speech sounds just as clear at 128 kbps mono.

For pure speech, 128 kbps mono is a well-established podcast-industry standard. A music intro can be worth 192 kbps stereo if mixed in.

Music track for archive

InputWAV stereo 44.1 kHz 16-bit, 4 min = 40 MB, bitrate 320 kbps stereo

OutputMP3 approx. 9.6 MB (4x smaller). Near-transparent quality against the original.

320 kbps is the highest MP3 bitrate and gives the least audible quality loss. For lossless archives use FLAC.

Common use cases

Podcasters converting recordings to MP3 for upload to hosting services.
Musicians sharing demo tracks by email or streaming platforms.
Students compressing lecture recordings for storage on mobile devices.
Project workers creating audio attachments for reports and presentations.
Developers testing how MP3 at various bitrates affects file size and quality.

Frequently asked questions

Why only WAV as input?: WAV contains raw PCM audio data that can be read directly by the browser’s `AudioContext.decodeAudioData`. The format supports 16-bit, 24-bit and 32-bit float without quality loss. Other formats like AAC, OGG, FLAC or M4A would require bundling separate decoders. To convert between compressed formats, use the original source or a desktop tool like ffmpeg.
What is the difference between mono and stereo?: Mono has one audio channel, stereo has two (left and right). For pure speech, stereo gives no benefit and needlessly doubles file size. Mono is the standard for radio news, single-voice podcasts and telephony. Stereo matters when there is spatial information such as instruments panned wide, vocals with panning, or nature field recordings. "Keep original" preserves the same channel count as the WAV file.
Is MP3 still a good format in 2026?: MP3 is over 30 years old but still the most compatible audio format on earth: every browser, media player, car, DJ setup and streaming platform supports it. Patents expired in 2017, so the format is now free. Newer formats like AAC (slightly better quality at low bitrates), Opus (best for voice and streaming) and FLAC (lossless) offer technically superior performance but lack MP3’s universal reach. For maximum compatibility, pick MP3; for better quality at the same size, pick AAC or Opus.
Are my files sent to a server?: No. Both WAV decoding (browser AudioContext) and MP3 encoding (LAME/lamejs in a Web Worker) run one hundred per cent locally on your device. Once the page has loaded you can turn off the internet and the tool still works. No file bytes leave your computer during upload or download. You can confirm this in the developer tools by inspecting the Network tab while converting.

Technical background

Process: the file is read as an `ArrayBuffer`, decoded with `AudioContext.decodeAudioData` into an `AudioBuffer` of raw PCM float32 data. A Web Worker receives the PCM and calls `lamejs.Mp3Encoder.encodeBuffer()` block by block. Progress is posted back via `postMessage`. Output is a `Blob` with MIME type `audio/mpeg` exposed through `URL.createObjectURL`. Bitrates: 128, 192, 256, 320 kbps CBR (constant bit rate). Channels: "keep" uses the WAV file’s native channel count, "mono" downmixes to one channel (average of L+R), "stereo" upmixes mono to two identical channels. Sample rate is preserved from the source (44.1 kHz, 48 kHz, etc.). No network calls after initial load.