Extract Audio from WebM to AU — Free Online Tool
Extract audio from a WebM file and save it as a Sun AU file, converting the Opus or Vorbis audio stream to uncompressed PCM (16-bit big-endian) in the process. This is useful for Unix/Linux workflows or legacy systems that require the simple, headerless AU format rather than modern compressed audio containers.
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FFmpeg Command
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DownloadHow It Works
WebM files typically carry audio encoded with either Opus or Vorbis — both lossy compressed codecs. The AU format, developed by Sun Microsystems, does not support these codecs; it is built around raw PCM audio. During this conversion, FFmpeg discards the video stream entirely (the WebM file may contain VP9 video), then decodes the Opus or Vorbis audio to raw PCM samples, and re-encodes those samples as 16-bit big-endian PCM (pcm_s16be) written into the AU container. This means the output is uncompressed, so file sizes will be significantly larger than the original WebM audio, but the audio data is now in a universally readable raw format with no lossy codec layered on top.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles all decoding, stream selection, and re-encoding in this conversion. |
-i input.webm
|
Specifies the input WebM file. FFmpeg reads the Matroska-based WebM container and identifies its streams — typically a VP9 video track and an Opus or Vorbis audio track. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the VP9 video stream in the WebM file. Since AU is a pure audio format with no video support, this flag is required to avoid an error. |
-c:a pcm_s16be
|
Decodes the Opus or Vorbis audio from the WebM and re-encodes it as 16-bit signed big-endian PCM — the default and most compatible encoding for the Sun AU format, matching the byte order of the SPARC architecture AU was designed for. |
output.au
|
Sets the output filename and tells FFmpeg to write a Sun AU container. FFmpeg uses the .au extension to select the correct muxer, which writes the AU magic number header followed by the raw PCM audio data. |
Common Use Cases
- Supplying audio to legacy Unix or Solaris workstation software that only reads Sun AU files and cannot handle modern WebM or Opus streams
- Extracting the narration or voiceover track from a WebM screen recording for ingestion into older audio editing tools that support AU but not Opus
- Converting WebM audio to a raw PCM format for signal processing pipelines or academic research scripts written for Unix systems that expect AU-formatted input
- Archiving audio from WebM web video into an uncompressed, codec-agnostic format that does not depend on Opus or Vorbis decoder availability
- Stripping compressed audio from a WebM file and preparing it as a PCM AU source before re-encoding to a different target format in a multi-step workflow
- Generating AU audio samples from WebM recordings for use in Unix system sounds or early internet audio applications that were originally designed around the AU format
Frequently Asked Questions
There is one unavoidable quality loss: the original Opus or Vorbis audio in the WebM file is lossy-compressed, and that compression introduced distortion when the WebM was first created. Decoding it to PCM for the AU file does not restore that lost quality. However, the PCM output in the AU file is itself lossless — no additional degradation is introduced by the conversion. What you get is an uncompressed representation of what the Opus or Vorbis decoder produces.
WebM stores audio using Opus or Vorbis compression, which can achieve very small file sizes — Opus at 128 kbps is perceptually transparent for most content. The AU format stores audio as raw 16-bit PCM samples with no compression at all. A one-minute stereo audio track at 44.1 kHz will occupy roughly 10 MB as uncompressed PCM, compared to under 1 MB as Opus. The size increase is expected and is inherent to the uncompressed nature of the AU format.
No. The Sun AU format has an extremely minimal header — it stores only basic technical parameters like sample rate, channel count, and encoding type. It has no standardized fields for metadata such as title, artist, album, or track number. Any metadata embedded in the WebM file will be lost during this conversion, so if preserving tags matters, you should export them separately before converting.
The AU format supports several PCM encodings including 8-bit signed (pcm_s8), 8-bit unsigned (pcm_u8), 16-bit signed big-endian (pcm_s16be), and the telephony codecs G.711 A-law and mu-law. FFmpeg defaults to pcm_s16be because it offers a good balance of dynamic range and broad compatibility — 16-bit PCM is the standard CD-quality depth, and big-endian byte order is native to the Sun/SPARC architecture the format was designed for. If you need a different encoding, you can change -c:a in the command.
Yes. Replace -c:a pcm_s16be with another supported AU codec such as -c:a pcm_mulaw for G.711 mu-law (used in telephony, 8-bit), -c:a pcm_alaw for G.711 A-law, or -c:a pcm_s8 for 8-bit signed PCM. For example: ffmpeg -i input.webm -vn -c:a pcm_mulaw output.au. Note that mu-law and A-law reduce audio fidelity significantly and are intended for voice, not music.
The displayed command processes a single file, but you can batch process on the command line using a shell loop. On Linux or macOS: for f in *.webm; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.webm}.au"; done. On Windows Command Prompt: for %f in (*.webm) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.au". This runs the same conversion on every WebM file in the current directory.
Technical Notes
The Sun AU format is one of the oldest digital audio container formats still recognized by modern software, originally developed for Sun Microsystems workstations and the NeXT computer. Its header is minimal: a 24-byte magic number block followed by fields for data offset, data size, encoding type, sample rate, and channel count. Because there is no compression and no complex codec negotiation, AU files produced by FFmpeg are immediately readable by virtually any Unix audio library. One important limitation is that AU does not support multiple audio tracks — if the WebM source contains multiple audio streams (e.g., multiple language tracks), only the default stream will be extracted. Additionally, the AU format's big-endian byte order (pcm_s16be) means the files are not natively efficient on little-endian x86 systems, though all modern software handles the byte-swap transparently. Sample rates and channel counts from the WebM source are preserved in the AU header, so a stereo 48 kHz WebM (the native sample rate for Opus) will produce a stereo 48 kHz AU file. No resampling occurs unless you explicitly add -ar to the command.