Audio Converter
Pull the audio track out of a video, or convert an audio file from one format to another, without uploading anything. Drop in an MP4, MOV, MKV, WebM, or AVI recording — or a WAV, FLAC, M4A, OGG, or AAC file — then pick the output format, bitrate, sample rate, and channel layout. The converter runs FFmpeg compiled to WebAssembly inside the browser tab, so a two-hour meeting recording is decoded on your own machine instead of being sent to a stranger's server and queued behind other people's uploads. The engine is only fetched once you actually choose a file, and it is cached afterwards.
- Video or audio in — MP3, WAV, AAC, FLAC, or OGG Vorbis out
- Nothing is uploaded: decoding and re-encoding happen in your browser tab
- Control bitrate, sample rate, and mono/stereo instead of accepting a fixed preset
- Preview the result and see its size before you download
Drop a video or audio file here
The file stays on your machine — it is never uploaded
Reads MP4, MOV, MKV, WebM, AVI, MP3, WAV, FLAC, M4A, OGG, AAC, and most other formats FFmpeg understands.
No file selected yet
Choose a video or audio file to begin. The conversion engine is only downloaded once you do.
Overview
One tool for two jobs that usually need two different websites: getting the sound out of a video, and moving an audio file between formats.
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Extract audio from video
Read an MP4, MOV, MKV, WebM, or AVI recording and write out just the sound. The video stream is dropped, so a 400 MB screen recording becomes a small audio file you can actually email or transcribe.
- 02
Convert between audio formats
Go from WAV to MP3 to shrink a recording, from FLAC to MP3 for a device that will not play lossless, or from M4A to WAV when an editor demands uncompressed input.
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Five output formats
MP3 for maximum compatibility, AAC for better sound quality at the same bitrate, OGG Vorbis when you want an open codec, and WAV or FLAC when nothing may be thrown away.
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Quality controls that matter
Set bitrate from 96 to 320 kbps, resample to 22.05, 44.1, or 48 kHz, and downmix to mono — which alone halves the size of a voice recording with no audible loss.
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Local processing
FFmpeg is compiled to WebAssembly and runs in the tab. Nothing is uploaded, there is no size cap imposed by a server, and no account is required.
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Preview before downloading
The result is playable in place and its size is shown next to the original, so you can hear whether 128 kbps is good enough before committing to the download.
How to use
From a video file to a finished audio file in four steps.
- 01
Drop a video or audio file onto the page, or use the file picker. The conversion engine starts downloading at this point — it is about 31 MB and only happens once.
- 02
Check the source summary: duration and the detected audio codec confirm the file was read correctly.
- 03
Pick an output format. Adjust bitrate, sample rate, and channels if the defaults do not suit — mono at 96 kbps is plenty for speech, while music is better at 192 kbps or above.
- 04
Press Convert, wait for the progress bar, then play the result to check it and download.
Details
Details that decide whether a conversion is usable.
- Accepts video and audio input, so one page covers both extracting and converting
- MP3, WAV, AAC/M4A, FLAC, and OGG Vorbis output
- Bitrate presets from 96 to 320 kbps for the lossy formats
- Resampling to 22.05, 44.1, or 48 kHz, or keep the source rate
- Stereo and mono downmixing
- Duration and source codec detected and shown before conversion
- Real byte-level progress while the engine downloads, not a fake animated bar
- In-page playback of the result before download
- Output size shown as a percentage of the original
- Conversion can be cancelled while it is running
Use cases
The situations where doing this locally is meaningfully better than uploading.
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Meeting and interview recordings
A recorded call often contains information you are not free to upload to a third-party converter. Extracting the audio locally keeps the recording inside your own machine while still giving you a small file to archive or send to a transcription tool you have vetted.
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Lecture and course video
A two-hour lecture video is mostly a static slide. Dropping the video stream turns a multi-gigabyte file into an audio file small enough to keep on a phone for the commute.
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Preparing audio for editing
Some editors and DAWs only accept WAV. Converting an M4A voice memo first means the session works from PCM — it cannot restore what the M4A already discarded, but it satisfies tools that refuse compressed input.
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Shrinking a voice recording
A WAV field recording is enormous for what it contains. Converting to mono MP3 at 96 kbps typically cuts it by more than 90% with no audible difference for speech.
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Playing a file on a device that refuses it
Older car stereos, cheap MP3 players, and some TVs only accept MP3. Converting FLAC or OGG Vorbis to MP3 is often the shortest path to a file that simply plays.
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Meeting a platform's format requirement
Podcast hosts, upload forms, and transcription services often accept a narrow list of formats and reject everything else. Converting to a plain MP3 or WAV first is usually quicker than arguing with an upload validator.
See also
Media files are not the only thing processed locally here. To shrink a screenshot before sharing it, use Image Compressor. To open a download your operating system refuses to unpack, there is Archive Extractor. And to check what a file quietly carries with it before you publish it, see Image Metadata Viewer.
Best practices
Choices that keep quality where you want it and file size where you need it.
- For speech, mono at 96–128 kbps is usually indistinguishable from the original and roughly a quarter of the size
- For music, stay at 192 kbps or above in MP3 — below that, cymbals and reverb tails start to smear audibly
- Prefer AAC over MP3 when the playback device supports it: it sounds better than MP3 at the same bitrate
- Do not convert a lossy file to another lossy format more than necessary — each pass discards more, and the loss never comes back
- Convert to WAV or FLAC when the file is going into an editor, and only compress at the end of the chain
- Leave sample rate on "Keep original" unless a target system specifically demands a different rate — resampling cannot add detail back
- Run long recordings on a desktop browser; phones have far less memory available to a tab
Limitations
What this tool deliberately does not do, and where it will struggle.
- The engine is about 31 MB on first use. It is cached afterwards, but the first conversion in a fresh browser profile carries that download.
- WebAssembly is 32-bit and the input and output both live in memory, so very large files — roughly beyond a few hundred megabytes — can exhaust the tab.
- Mobile browsers give a tab much less memory than a desktop one and will fail earlier on long recordings.
- Conversion is single-threaded here. That is deliberate: multi-threading requires cross-origin isolation, which would break other parts of the page, and audio encoders barely benefit from it anyway.
- This converts, it does not edit. Trimming, merging, changing volume, and mixing are outside its scope.
- The video codec never matters here: the picture is dropped before anything is decoded, so an AV1 or HEVC recording is no harder than an H.264 one. Only a file whose audio codec the engine does not include will fail to read.
- DRM-protected files cannot be read, in this tool or any other that does not break the protection.
FAQ
Questions that come up when converting media in a browser.
How do I convert MP4 to MP3?
Drop the MP4 onto the page, leave the format on MP3, and press Convert. The video stream is discarded and only the audio track is encoded, so the output is a fraction of the original size. You can adjust bitrate first if you want a smaller file or higher quality.
Is my video uploaded anywhere?
No. The conversion engine is FFmpeg compiled to WebAssembly, and it runs inside your browser tab. The file is read from disk into the tab's memory and the result is written back out as a download. Nothing about the file is sent over the network — you can confirm this by watching the network tab while converting.
Which bitrate should I choose?
For speech, 96–128 kbps in mono is usually indistinguishable from the source. For music, 192 kbps is a reasonable floor and 256–320 kbps is close to transparent for most listeners. Higher bitrates cost size without adding detail that was not in the source to begin with.
Why does the first conversion take so long to start?
The conversion engine is roughly 31 MB and is downloaded the first time you select a file. It is cached with a long expiry afterwards, so later visits and later conversions start immediately. The download is deliberately deferred until you actually pick a file, so simply visiting the page costs you nothing.
What is the difference between MP3, AAC, FLAC, and OGG Vorbis?
MP3 is lossy and plays on essentially everything. AAC is also lossy but sounds better than MP3 at the same bitrate. OGG Vorbis is comparable to AAC and is an open format that generally carries no licensing cost, though fewer consumer devices accept it. FLAC is lossless — it compresses without discarding anything, at roughly half the size of WAV.
Does converting MP3 to MP3 reduce quality?
Yes. Every lossy encode discards information permanently, so re-encoding an MP3 as another MP3 loses a little more even at a higher bitrate. If you need a different bitrate, always re-encode from the original source file rather than from an already-compressed copy.
What is the largest file I can convert?
There is no server-imposed limit, but WebAssembly is 32-bit and the input and output share one memory space. In practice files beyond a few hundred megabytes become unreliable, and the tool warns you when a file crosses that threshold. Desktop browsers handle considerably more than mobile ones.
Can I use this on my phone?
Short files usually work. Long recordings often do not — mobile browsers allocate a tab far less memory than desktop ones, and the tab is killed when the limit is reached. For anything over a few minutes, a desktop browser is the safer choice.
Why is my FLAC file larger than the MP3 it came from?
FLAC is lossless, so it stores everything the input contained. Converting an MP3 to FLAC does not restore the detail MP3 threw away — it simply stores the already-degraded audio without further loss, in a larger file. Converting lossy to lossless is only worth it when a specific tool demands that format.
Can I extract audio without re-encoding it?
Not in this tool. Copying the audio stream untouched is possible in principle, but it only works when the source codec and the target container agree, which is a confusing failure mode to expose. Re-encoding always produces a file that plays, which is the more useful default here.
Related tools
More local, browser-only tools for working with files.