WAV to MP3 and AAC with the timecode still on it

Drop your WAV files onto the box above, or click it to browse, or use the folder button to add an entire card. Leave the output format on .mp3, pick a bitrate, and press Convert all. The MP3s are ready in seconds, and a .zip of the batch downloads by itself once the last file finishes. Nothing is uploaded and there is no sign-up.

Exactly the same way, except you set the output format to .aac before you convert. The .aac file is a real ADTS AAC-LC stream, so the extension matches what is actually inside it and every transcription service and ffmpeg-based pipeline accepts it as is. AAC encoding uses the browser's built-in encoder, so .aac output needs Chrome, Edge, or Safari 26 and later. On an older Safari or on Firefox the .aac option switches itself off and you use MP3, which every service accepts anyway.

It is 100% free, forever. No account, no sign-up, no email, no file limit, no watermark, no paid tier at the end. The browser version, the installed app and the single file download are the same converter with every feature switched on. It runs entirely on your machine, so it costs me nothing to leave up. If it saves your day on a job, tell another sound person about it.

No. Every conversion runs on your own machine, inside your browser. Your audio is never uploaded, never stored and never seen by anyone else, so there is no waiting on a multi-gigabyte WAV to crawl up to somebody's server and confidential dailies never leave the room.

Yes, a little, because MP3 and AAC are lossy formats: they throw away data your ears are least likely to miss in order to make the file roughly ten times smaller. For transcription, editorial review and client copies that difference does not matter at all. For the finished mix or anything going back into a conform, keep the WAV. That is why this tool never touches your original.

128 kbps is plenty for speech and gives you a file about one tenth the size of the WAV. Go to 192 or 256 kbps if the audio is noisy, heavily accented, multi-speaker or a crowded room, and drop to 96 kbps only when you are fighting a hard upload limit. Bitrates from 96 to 320 kbps are available here.

Never. Source files are opened read-only through the browser's file APIs and every output is a brand new file written next to nothing. The master WAV on your card or drive is untouched, byte for byte, and stays the master for conform and sync.

Yes, and that is the whole point of this tool. The BWF TimeReference value is read out of the bext chunk and converted to real start timecode at the file's own frame rate, then written into the output file's tags and into an optional conversion report. Supported rates are 23.976, 24, 25, 29.97 drop and non-drop, 30, 48, 50, 59.94 drop and non-drop, and 60. Drop-frame math is done properly.

Start timecode, frame rate and the raw sample count, scene, take, tape and project, track names, originator and origination date, the mixdown that was used, and a SHA-256 hash of both the source and the output. It goes into the output file's own tags, and optionally into a .txt report you can read and a .json file a machine can read. That paperwork is the point: nobody can claim the timecode or the scene and take got lost in the handoff.

Yes. Use the folder button, or drag a whole folder onto the drop zone. Subfolders are scanned, non-WAV files are ignored, duplicates are skipped, and the batch converts in order with an overall progress bar. When the last file lands, a .zip of everything downloads automatically.

There is no size limit. Files are read in blocks rather than loaded whole, so multi-gigabyte poly recordings are fine, along with RF64 and BW64 files that are already past the 4 GB WAV ceiling. Files over 512 MB skip the source SHA-256 only, because of a browser hashing limit, and the report says so.

By default you get the original L/R stereo mix exactly as recorded, or a mono mix folded down from those same two tracks, both taken from tracks 1 and 2, the mixer's own mix. Isolated tracks are never summed into those two, because that doubles up dialogue and gives a transcription service a worse file than the mix does. Drop in a poly file and two more choices appear: a mono mix of every track, or Choose tracks, which lists the real channels in the files you loaded, labelled with their iXML track names, so you can export the boom on its own or a single lav. Tick one track and you get a mono file, tick two and you get a stereo pair in that order, tick three or more and they are summed to mono. Whichever you chose is written into the tags and the report.

Either works. Every major service, Rev, Otter, Trint, Descript, Sonix and anything built on ffmpeg or Whisper, takes both. MP3 is the safest default and works in every browser. AAC is smaller at the same quality, which helps if you are uploading a long interview over a hotel connection or a hotspot.

It is made for a computer: Chrome, Edge, Firefox or Safari on Mac, Windows or Linux. Chrome and Edge are the ones to use, because only they can save straight into a folder you pick, and that is what removes the limit on how big a batch can be. MP3 output works in every browser. AAC needs Chrome, Edge, or Safari 26 and later; on an older Safari the .aac option switches itself off and tells you why. There is no iPhone or iPad version. The page will load on one, but it runs very slowly, it cannot save into a folder, and it gives up after a few hours of audio, so it says so at the top and asks you to come back on a computer.

It is not offered as a phone or tablet tool, and you should not plan a day around one. The page will still load and convert on an iPhone or iPad, but expect it to be very slow, there is no folder picker in Safari so everything has to be held in memory and handed back as a zip, and a phone runs out of room after roughly two to three hours of audio at 128 kbps. On a computer in Chrome or Edge you pick a folder, each file is written to your disk and released the moment it finishes, and a full card converts in one pass. If a producer needs MP3s and all you have is a phone, it will get you a couple of files out of a hole. For anything more, use a laptop.

Yes, and it is worth setting up before you need it. Install it as an app from the card near the top of the page and it caches itself, so it opens and converts with no signal. Or download the single file version: one HTML file you can email, drop on a thumb drive or keep on set, that runs by double-clicking it on any Mac, Windows or Linux machine.

A broadcast WAV is an ordinary WAV with extra chunks bolted on, mainly bext and iXML, that carry the timecode, scene, take, project and track names post needs to sync and conform. Location sound recorders write them by default, which is why a shoot can wrap with nothing but WAVs on the card. When a producer then asks for MP3s for transcription, this is the tool that makes them without dropping the information that makes those files useful later.