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Supports: AV1, MP4, MKV, WEBM
AV1 is a video codec, so the job on this page is to pull the audio track out of an AV1 file and write it as uncompressed WAV (PCM) that any editor, DAW, or transcription tool opens without a decoder plug-in. The one thing worth understanding before you start is that AV1 itself codes no audio at all — sound rides in the container next to the video — so what you get depends on how your .av1 file was packaged. The steps below cover the normal case, and the walk-through explains the exception.
.av1 is not a standardised container extension, so two quite different kinds of file end up with that name. Knowing which one you have tells you immediately whether a WAV is even possible.
What your .av1 file really is |
Audio track? | What you get |
|---|---|---|
| AV1 video in a Matroska wrapper (the usual case, and what xconvert's own AV1 converter writes) | Yes — normally AAC or Opus | A normal WAV containing the full soundtrack |
| AV1 video in an MP4 or WebM wrapper, renamed | Yes — typically AAC in MP4, Opus in WebM | A normal WAV containing the full soundtrack |
A raw elementary bitstream (.obu, or an IVF dump) |
No | Nothing to extract; the source carries coded video frames only |
If you are unsure, the practical test is size and provenance: a file that came out of a converter, a download, or a re-encode is almost always a wrapped container with audio, whereas an elementary stream is something you had to deliberately demux out of one. When a file genuinely is video-only, the audio you are looking for lives in a separate file, and no tool can reconstruct it.
Every option below is uncompressed linear PCM inside a RIFF/WAV container, so the only real decisions are bit depth and how large you are willing for the file to be. The data rates are arithmetic, not estimates: sample rate multiplied by bit depth multiplied by channel count.
| Audio Codec option | Data rate, 44.1 kHz stereo | Roughly per minute | Use it when |
|---|---|---|---|
| PCM 16-bit Little Endian (default) | 1,411 kbit/s | about 10.6 MB | Anything normal — CD-standard depth, universally supported |
| PCM 24-bit Little Endian | 2,117 kbit/s | about 15.9 MB | You will apply gain, EQ, or heavy processing and want headroom |
| PCM 32-bit Little Endian | 2,822 kbit/s | about 21.2 MB | A mastering chain that specifically asks for 32-bit input |
| PCM A-law or mu-law | 706 kbit/s | about 5.3 MB | Telephony-style speech where an 8-bit companded file is acceptable |
If the file really is an elementary AV1 stream, there is no audio to recover here or anywhere else, and the fix is to find the original container or the separate audio file it was demuxed from. Damaged downloads are the other common dead end: a truncated container may still show a picture in a player while its audio index is unreadable. And if your source already sits in a normal container, the direct tools are often simpler — MKV to WAV and MP4 to WAV do the same extraction without the .av1 guesswork.
Usually yes, because most files named .av1 are AV1 video inside a Matroska, MP4, or WebM wrapper that also carries an audio track — typically AAC or Opus. The AV1 codec itself, published by the Alliance for Open Media in 2018, describes video only, so a genuine elementary bitstream has no sound in it at all.
Because the upload had no audio stream to decode. That happens with raw AV1 bitstreams, with files that were demuxed video-only, and occasionally with screen recordings captured without an audio device. Re-upload the complete container file and the soundtrack will come through.
PCM 16-bit little-endian, the CD-standard flavour that every editor and player accepts. Under Audio Codec you can switch to PCM 24-bit or 32-bit little-endian for editing headroom, or to PCM A-law or mu-law for compact 8-bit speech files. All of them are uncompressed linear PCM in a RIFF container.
The decode itself is exact: PCM stores the decoded samples with nothing thrown away. What WAV cannot do is undo the original lossy encode — if the container held Opus or AAC, the WAV is a faithful copy of that already-compressed signal rather than a restored master. That still matters in practice, because it means no further generation loss while you edit.
Audio Sample Rate offers Original plus 8000, 12000, 16000, 24000, 44100, and 48000 HZ; Audio Channel offers Original, Mono, and Stereo. Leaving both on Original avoids a needless resample, which is what you want unless a downstream tool demands a specific rate.
Only the format settings decide, never the content. Sixteen-bit 44.1 kHz stereo runs 1,411 kbit/s — about 10.6 MB per minute, or 635 MB per hour — and 48 kHz stereo is about 11.5 MB per minute. Halving the channels or dropping to A-law cuts that proportionally.
.av1 file that is really a Matroska container?Yes, and that is the common case. We tested it while writing this page: a short clip encoded as AV1 video plus AAC audio in a Matroska wrapper but saved with the .av1 extension is detected as Matroska, and its audio track extracts to PCM WAV cleanly. Detection is based on the file's contents, not its name.
WAV when the file is going into an editor, a DAW, or a transcription pipeline that wants raw samples. If the file is going to be emailed, uploaded, or kept on a phone, uncompressed PCM is wasteful — AV1 to MP3 or AV1 to M4A gives a fraction of the size for listening purposes.
Uploads and results are deleted automatically a few hours after conversion. Everything travels over an encrypted connection and is processed on our servers, with no account required, no watermark, and nothing shared or made public.