Audio Converter

An audio converter that runs in your browser

This audio converter turns MP3, WAV, M4A, AAC, OGG, Opus, FLAC and WebM files into MP3, WAV, M4A or OGG, and into FLAC where your browser can encode it. You can also add a video and keep only its sound. Add as many files as you like, choose the settings once, and convert them in one go. Each file shows what it contains before you start: the container and codec, sample rate, channels, length and bitrate.

Everything happens in your browser tab. There is no upload, no account and no queue on a server, so a long podcast or an unreleased song never leaves your computer or phone.

• Make a file play everywhere: convert an OGG or FLAC file to MP3 for an old car stereo, a website or a messenger that only takes MP3.
• Prepare audio for editing: convert compressed files to WAV for an editor that wants uncompressed sound.
• Save space: turn large WAV recordings into M4A or MP3 at a bitrate that suits them, or into mono for speech.
• Match a required format: some platforms ask for 44.1 or 48 kHz; the sample rate setting converts to either.

Which format to choose

MP3, AAC and Opus are lossy: they throw away detail that is hard to hear, which makes the files small. WAV stores the sound uncompressed, and FLAC compresses it without losing anything further, as the MDN guide to audio codecs explains. This tool always writes FLAC with 16-bit samples, so its FLAC files are lossless for 16-bit sources; 24-bit or float audio is reduced to 16-bit. Neither WAV nor FLAC can restore detail that a lossy source already lost. Converting an MP3 to WAV or FLAC only makes it bigger. Converting from one lossy format to another re-encodes the sound and adds a little more loss each time, so convert from the best copy you have.

Bitrate, sample rate and channels

Bitrate sets how many kilobits per second a lossy file may use. The size follows directly: a minute of MP3 at 128 kbps takes about 0.92 MB, at 192 kbps about 1.37 MB and at 320 kbps about 2.29 MB. The MDN guide suggests at least 192 kbps for music in MP3, and lists 96 kbps at 48 kHz as the minimum for stereo AAC and Opus. MP3 is written at a constant bitrate. For AAC and Opus the bitrate is an average that the browser's encoder aims for, with a variable bitrate by default (WebCodecs specification), so those files can come out a little smaller or larger than the estimate.

WAV bit depth sets the size of uncompressed files. A minute of 16-bit stereo needs about 10.09 MB at 44.1 kHz and 10.99 MB at 48 kHz; 24-bit needs half as much again, and 32-bit float twice as much as 16-bit. Choose 24-bit or 32-bit float when the file goes into an editor where you will change the volume or add effects. 16-bit reduces a 24-bit or float source to 16-bit.

Sample rate: Keep leaves the rate as it is where the format allows it. This tool writes MP3 in the MPEG-1 mode, which MDN describes as the better one for music and which stores 32, 44.1 or 48 kHz, so a 22.05 kHz recording becomes 44.1 kHz, and the file row tells you. OGG files are always written at 48 kHz, because Opus does not preserve the original rate (RFC 7845). When the rate changes, the tool resamples with a band-limited filter that removes everything above the new limit first, so high tones do not fold back as noise.

Channels: Mono mixes left and right into one channel and suits speech. Stereo turns a mono file into two identical channels for players or platforms that expect stereo. Surround files with more than two channels are always mixed down to stereo.

On iPhone, iPad and older browsers

M4A, OGG and FLAC are encoded by the browser through the WebCodecs AudioEncoder. According to MDN's compatibility data, it arrived in Chrome and Edge 94, Firefox 130 on computers and Safari 26, including Safari on iPhone and iPad; Firefox for Android does not have it. Without it, the tool offers MP3 and WAV only and says so above the file list. Even with WebCodecs, browsers and operating systems support different codecs, so the tool asks your browser which formats it can write and hides the others.

Reading files works the same way: where the browser has the WebCodecs AudioDecoder, files are read in small blocks, which needs little memory. Otherwise the tool falls back to the Web Audio decoder, which decodes the whole file at once and needs more memory, so very long files (over 90 minutes or 1.5 GB) are turned away there. Phones and tablets have less memory for a browser tab, so there the limit is 30 minutes or 400 MB. Which input codecs open at all depends on the decoders in your browser; if a file does not open, try another browser. Tags are not kept for a file that only the browser can read.

How to convert audio files

1. 1Drop one or more audio or video files on the box, or tap it to choose them. Each file shows its format details.
2. 2Pick the format, then the bitrate or bit depth, the sample rate and the channels. The estimated output size updates as you change them.
3. 3Press Convert. Every file shows its own progress bar, and Cancel stops the conversion.
4. 4Play a result to check it, then download files one by one or all together as a ZIP.

Need a different job? Trim Audio cuts a clip out of a file, the MP4 to MP3 converter is made for single videos, and Extract Audio from Video saves a video's sound as WAV.

Open-source libraries do the work: Mediabunny (MPL-2.0) reads and writes the files and their tags, lamejs (LGPL-3.0), a JavaScript port of the LAME encoder, writes the MP3 files, and fflate (MIT) builds the ZIP.

Built by Tobias Kurth, designer. Updated October 14, 2026.

Frequently asked questions