Audio Converter
Convert audio, WMA, AMR and AC-3 included, to MP3, WAV, FLAC, AIFF, M4A or OGG in your browser — trim, fade, normalise and resample, without uploading the file.
🔒 Decoding and encoding both happen in this page — nothing is uploaded. Output is MP3, M4A (AAC), OGG (Opus), FLAC, WAV or AIFF. AAC and Opus come from your browser’s own encoder and the other four from code on this page; the containers are written here either way. The audio is re-encoded from decoded samples, so cover art and tags are not carried over.
Drop audio or video files here or click to browse
Convert audio without uploading a thing
Drop an audio file, pick an output format, and download the result. Decoding starts with the Web Audio API — your browser's own audio engine — so it opens whatever your browser can already play: almost always MP3 and WAV, usually M4A/AAC, and often OGG and FLAC. When the browser refuses a WMA, AMR, AC-3, AU or VOC file, the page decodes it itself: AU and VOC with its own code, and WMA, AMR and AC-3 with a decoder fetched from this site the first time a file of that kind is opened — opencore-amr for AMR, and FFmpeg's own decoders, under the LGPL, for WMA and AC-3. Every step runs on your device, so there is no ffmpeg to install and nothing to wait on a server for: the only parts of FFmpeg here are its WMA and AC-3 decoders, and they are fetched only when such a file is opened.
Six output formats, and what each is for
WAV and AIFF are uncompressed 16-bit PCM: lossless, and about 10 MB a minute in stereo. FLAC is lossless too and smaller — by how much depends on the recording. MP3 is the one everything plays, at any bitrate the format allows — 32 to 320 kbps for 32/44.1/48 kHz sources, 8 to 160 for the half-rate MPEG-2 range. M4A holds AAC and OGG holds Opus; both are encoded by your browser itself through WebCodecs, so no encoder is downloaded — and if your browser has no encoder for the one you picked, the page says so and names one that does rather than quietly writing something else.
Editing, and the limits it keeps
Before re-encoding you can trim the start and end, fade in and out, reverse, force mono or stereo, set the volume or normalise the peak or the average level, and change the sample rate through a windowed-sinc resampler. For MP3, M4A and OGG, "Aim at a file size" in the bitrate select takes a "Target size (MB)" and picks the highest bitrate whose audio fits in it — tags and the container add a little on top, and a target no bitrate can reach is refused. Drop several files and they share one set of settings and come back as one ZIP. The format, bitrate, sample rate and channels are remembered on this device until you press Forget saved settings. Under Tags (optional) you can write a Title, Artist, Album, Year and Genre into the file — an ID3 tag in MP3, comments in FLAC and OGG, an INFO chunk in WAV; M4A and AIFF are written without tags, and the page says so when you pick one. What the tool cannot do exactly, it refuses: Opus holds only 8, 12, 16, 24 and 48 kHz, so a 44.1 kHz source is refused with those rates named rather than resampled behind your back, and AAC runs a few milliseconds longer than its source because it codes 1,024 samples at a time — the page says so on screen instead of hiding it. There is no mixing, and a video file gives up only its sound — from the containers your browser can open, usually MP4 and WebM.
Frequently asked questions
Which input formats can I convert?
Whatever your browser can decode through the Web Audio API — almost always MP3 and WAV, usually M4A/AAC, and often OGG and FLAC — plus WMA, AMR, AC-3, AU and VOC, which this page decodes itself when the browser cannot. Support varies by browser and operating system, so if a file still can't be decoded you get a clear error instead of a broken download.
Is my audio uploaded to a server?
No. Decoding uses your browser's built-in audio engine — or, for a WMA, AMR, AC-3, AU or VOC file it cannot open, this page's own decoder running on your device — and the encoding runs on your device too, so the file never leaves your computer. Even private recordings stay local, and once it has been used it keeps working offline for any file your browser decodes itself. A WMA, AMR or AC-3 file needs the connection once more the first time a file of that kind is opened, to fetch its decoder.
Should I choose WAV or MP3?
WAV is uncompressed 16-bit PCM: lossless but big, roughly 10 MB per minute in stereo. MP3 is much smaller, at a bitrate you choose, such as 128, 192 or 320 kbps. Pick WAV for editing or archiving, and MP3 for sharing or uploading.
Can I add a title, artist or album to the converted file?
Yes. Fill in any of Title, Artist, Album, Year and Genre under “Tags (optional)”, and they are written where each format’s players look: an ID3 tag in MP3, a Vorbis comment block in FLAC, the comment header in OGG, and a LIST/INFO chunk in WAV. Every file of a batch gets the same tags. M4A and AIFF are written without tags — pick one of those and the fields switch off with a line saying so. Left empty, the fields change nothing in the file.
What happens when my browser cannot open a WMA or AMR file?
The page decodes it instead. Your browser is asked first; when it refuses a WMA, AMR or AC-3 file, the page fetches the decoder for that format from this site — FFmpeg's own WMA or AC-3 decoder, or opencore-amr for AMR — and shows “Loading the decoder for this format” with its size while it arrives. That download happens the first time a file of that kind is opened, and the decoder runs on your device, so the file itself is still not uploaded. AU and VOC need no decoder: the page reads them with its own code. The decoded audio then goes through the same trim, level and format options as any other file. DSS dictation files do not open.
Can I aim at a file size instead of a bitrate?
Yes, for MP3, M4A and OGG. Pick "Aim at a file size" in the bitrate select and type a "Target size (MB)": each file gets the highest bitrate whose audio fits in it over its length after any trim, and the option names the bitrate it chose. Tags and the container add a little on top, and a target no bitrate can reach is refused on screen.
Can it take the vocals out of a song?
No. Taking a voice out of a mix is source separation, a job for a trained neural model, and this converter does not carry one. The shortcut some tools use — subtracting one stereo channel from the other — also removes the bass and drums mixed to the centre and leaves the voice’s reverb behind, so it is not offered under that name. What the converter does change is the trim, the channels, the sample rate, the level, the fades and the direction of play.