EZ
EZ2Conv

Audio Converter

Convert audio files between MP3 and WAV formats with adjustable bitrate, sample rate, and mono/stereo channel settings. All processing runs in your browser -- your audio is never uploaded to any server, ensuring total privacy.

Drop audio file here or click to upload

Supported formats: .mp3, .wav, .ogg, .aac, .m4a, .flac, .webm

All processing happens in your browser. Your audio never leaves your device.

How to Use Audio Converter

  1. Upload your audio file (MP3, WAV, OGG, FLAC, AAC, M4A up to 500MB) by dragging and dropping or clicking the upload area
  2. Choose your output format (MP3 for smaller files or WAV for lossless quality), then set bitrate, sample rate, and mono or stereo channels
  3. Click Convert to begin processing -- the tool decodes, re-encodes, and optimizes your audio entirely within your browser
  4. Preview the converted audio with the built-in player to verify quality, then download the file when satisfied with the result

Tips

  • MP3 offers smaller file sizes, ideal for music and podcasts
  • WAV provides lossless quality, perfect for professional audio editing
  • Higher bitrate means better quality but larger file size
  • Use Mono for voice recordings to reduce file size

How This Audio Converter Works

What this converter does

This tool takes an audio file and re-encodes it into either MP3 or WAV. It accepts a broad range of inputs — MP3, WAV, OGG, AAC, M4A, FLAC, and WebM — and decodes them through the Web Audio API, which turns the compressed source into raw audio samples. From those it writes a fresh file in the format you pick, applying your chosen sample rate and channel layout.

MP3 output is produced by a LAME encoder (lamejs) running as JavaScript, while WAV is written as uncompressed 16-bit PCM. Files up to 500MB are accepted, and the decode-and-encode pass runs on your own device without an upload.

When to use it

Reach for it when a file is in a format your target app or device will not play, or when you want a smaller, more portable copy. A frequent case is converting voice memos or downloaded tracks that arrive as M4A or FLAC into MP3, which nearly every player and phone handles.

It is just as useful in the other direction: exporting a WAV from a compressed source when you need an uncompressed file for editing, or collapsing a stereo track to mono to shrink a spoken-word recording. Tuning bitrate or sample rate to hit a size or quality target is part of the same step.

Example: an M4A voice memo to MP3

Say you have a lecture recorded as a 40MB M4A file. Choose MP3 as the output, set the bitrate to 128 kbps, and switch channels to mono, since speech gains nothing from stereo. The Web Audio API decodes the AAC audio inside the M4A, the LAME encoder re-encodes it at 128 kbps, and the result is a far smaller MP3 that plays anywhere. An estimated size appears before you commit, and the built-in player lets you check it first.

Formats, quality, and edge cases

Output is limited to MP3 and WAV, which cover the compressed-and-portable versus uncompressed-and-editable split most workflows need. The bitrate options (128, 192, 256, and 320 kbps) apply only to MP3; WAV size is governed by sample rate and channel count. Sample rate can be 44.1 kHz (CD standard) or 48 kHz (the usual choice for video).

A few limits are worth knowing. Choosing mono averages the left and right channels, so it cannot be undone. Very large files can exhaust available memory during decoding, so a shorter clip works better. And if a source uses a codec the browser cannot decode, converting it to a common format first usually clears the problem.

Frequently Asked Questions

Every conversion is free, including the full 320 kbps bitrate and files up to the 500MB limit. No premium tier gates the higher quality settings, and all output formats and mono/stereo options are available from the first use — there is nothing to subscribe to.
Yes. Decoding runs through your browser's Web Audio API and MP3 encoding is handled by a JavaScript LAME encoder (lamejs), both executing on your device. The audio file itself is never uploaded, so recordings, music, and other sensitive tracks stay with you.
No. Drop in a file, pick MP3 or WAV, adjust the bitrate and channels, then download — there is no sign-up, email step, or registration standing between you and the converted file.
You can import MP3, WAV, OGG, AAC, M4A, FLAC, and WebM, and export to either MP3 or WAV. So an M4A or FLAC file can become an MP3, or a compressed track can be exported as an uncompressed WAV for editing. Output is intentionally kept to those two widely supported formats.
For music, 192–320 kbps MP3 keeps quality high; for podcasts or voice, 128 kbps in mono is plenty and much smaller. Sample rate can stay at 44.1 kHz for general listening or 48 kHz to match video. Bitrate applies to MP3 only, since WAV is stored uncompressed.
Size depends on the target format and settings, not the source. WAV is uncompressed, so it is usually much larger than an MP3 or M4A input, while converting to MP3 at a lower bitrate shrinks the file and a higher bitrate grows it. An estimated size is shown before you convert so there are no surprises.