Convert M2TS to AU — Free Online Tool
Convert M2TS Blu-ray video files to AU audio format, extracting the audio stream and encoding it as 16-bit big-endian PCM — the native Sun AU codec. This is the go-to tool for pulling lossless-quality raw audio from Blu-ray or AVCHD camcorder footage into a Unix-compatible uncompressed audio file.
to
FFmpeg Command
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Note: Browser-based encoding uses approximate quality targets. For precise CRF compression, copy the FFmpeg command above and run it on your desktop.
Estimated output:
Conversion Complete!
DownloadHow It Works
M2TS files are MPEG-2 Transport Stream containers typically carrying H.264 or H.265 video alongside multi-channel audio encoded as AAC, AC-3, DTS, or TrueHD. The AU format is a simple audio-only container developed by Sun Microsystems that supports only PCM audio codecs — it cannot hold video, subtitles, chapters, or multiple audio tracks. During this conversion, FFmpeg discards the video stream entirely and selects the first audio track from the M2TS file, transcoding it to 16-bit big-endian signed PCM (pcm_s16be) — the default and most compatible codec for AU files. Because AU is uncompressed PCM, the resulting file will be significantly larger than the original compressed audio stream, but represents a lossless decoded version of the source audio at its original sample rate and channel layout.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool. In this browser-based tool, this runs as FFmpeg.wasm compiled to WebAssembly — the same command works identically when run in a local terminal for files over 1GB. |
-i input.m2ts
|
Specifies the input file as an M2TS container — the BDAV MPEG-2 Transport Stream format used by Blu-ray discs and AVCHD camcorders. FFmpeg will parse all streams inside, including video, audio, and subtitles. |
-c:a pcm_s16be
|
Sets the audio codec to 16-bit signed big-endian PCM, which is the standard and most compatible codec for the Sun AU format. This decodes whatever compressed audio (AAC, AC-3, etc.) is in the M2TS and re-encodes it as uncompressed PCM samples in big-endian byte order. |
output.au
|
Specifies the output filename with the .au extension, telling FFmpeg to mux the PCM audio into a Sun AU container. Because AU is audio-only, FFmpeg automatically drops the video and subtitle streams from the M2TS without needing an explicit '-vn' flag. |
Common Use Cases
- Extracting the audio from an AVCHD camcorder recording (.m2ts) for use in Unix or Linux audio workflows that expect raw PCM in AU format
- Archiving the uncompressed audio from a Blu-ray rip for audio restoration or forensic analysis work on legacy Unix systems
- Feeding clean PCM audio from Blu-ray source material into old Sun workstation software or audio tools that only accept .au files
- Decoding compressed Blu-ray audio (AAC or AC-3) to uncompressed 16-bit PCM for use as a reference track when comparing audio codec quality
- Stripping and converting the audio layer of an AVCHD home video into a simple headerless-style AU file for use in early internet audio streaming pipelines
- Preparing raw audio from high-definition broadcast captures (.m2ts) for import into audio editors that support AU but not compressed transport stream audio
Frequently Asked Questions
It depends on the source audio codec in the M2TS file. If the M2TS contains lossless audio (such as TrueHD or PCM), the conversion to AU PCM is transparent with no quality loss. If the source contains lossy audio like AAC or AC-3 (the most common case on Blu-ray and AVCHD recordings), the AU file will be an uncompressed decode of that lossy stream — the audio is 'losslessly' copied into PCM, but the quality ceiling is already set by the original lossy encoding. No additional quality is lost in the conversion itself.
M2TS files store audio in compressed formats like AAC or AC-3, which can achieve 10:1 or greater compression ratios. AU with pcm_s16be is completely uncompressed — every sample is stored as raw 16-bit data. A 1-hour Blu-ray rip with stereo AAC audio at 192k might have an audio stream of roughly 80MB, but the equivalent uncompressed PCM in an AU file would be approximately 600MB or more depending on sample rate. The video stream is also dropped entirely, so the overall file size drop from the M2TS container is dramatic, but the audio portion itself expands significantly.
FFmpeg will decode and preserve the channel layout from the first audio track in the M2TS — so if the source is 5.1 surround, the AU file will contain 6 channels of PCM audio. However, AU is a very simple format with limited metadata support, and some legacy AU players and Unix tools may only expect mono or stereo files and could misinterpret multi-channel AU files. If compatibility with older software is a concern, you may want to downmix to stereo by adding '-ac 2' to the FFmpeg command.
All of them are discarded. AU is a pure audio format with no support for video streams, subtitle tracks, chapters, or multiple audio tracks. FFmpeg automatically selects only the first audio track from the M2TS and ignores everything else. If your M2TS has multiple audio tracks (e.g., a director's commentary alongside the main audio), only the default first track is extracted. To select a different audio track, you can add '-map 0:a:1' to the command to pick the second audio stream.
The command as shown preserves the original sample rate from the M2TS source. To resample the audio — for example, to convert a 48kHz Blu-ray audio track down to 44100Hz for compatibility with certain AU players — add '-ar 44100' before the output filename: 'ffmpeg -i input.m2ts -c:a pcm_s16be -ar 44100 output.au'. Most Blu-ray and AVCHD audio is natively 48kHz, so resampling may introduce a minor quality change and is only recommended when the target application requires a specific sample rate.
Yes. On Linux or macOS you can use a shell loop: 'for f in *.m2ts; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.m2ts}.au"; done'. On Windows Command Prompt, use: 'for %f in (*.m2ts) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.au"'. Each M2TS file will be processed sequentially, with the video discarded and the first audio stream converted to 16-bit big-endian PCM and saved as a matching .au file. This is especially useful when pulling audio from a large collection of AVCHD camcorder clips.
Technical Notes
The Sun AU format uses a minimal fixed-size header (24 bytes minimum) containing the audio encoding type, sample rate, channel count, and an optional annotation field. The pcm_s16be codec stores samples as 16-bit signed integers in big-endian byte order — a reflection of AU's origin on big-endian Sun SPARC hardware. This means AU files generated on little-endian x86 systems (including via FFmpeg.wasm in the browser) are byte-order correct for the format but may appear byte-swapped if opened naively as raw PCM without reading the header. AU supports pcm_s8, pcm_u8, pcm_alaw, and pcm_mulaw as alternative codecs if smaller file sizes or telephony compatibility are needed, though these come with significant quality tradeoffs. Notably, AU has no support for ID3 tags, Vorbis comments, or any modern metadata standard — artist, title, and album information from the M2TS container will be lost entirely. The format also lacks support for seeking in some implementations, which is why it retains a niche in streaming contexts. When processing M2TS files from AVCHD camcorders, be aware that the audio is almost always 48kHz stereo or 5.1 AC-3/AAC, and FFmpeg will faithfully decode and write all channels to the AU file unless explicitly remapped.