Extract Audio from MTS to AU — Free Online Tool
Extract audio from MTS camcorder footage and save it as a Sun AU file with uncompressed 16-bit big-endian PCM audio. This tool strips the H.264 video and AC-3/AAC audio from your AVCHD recording and converts it to a raw, lossless PCM stream in the classic Unix audio format — ideal for archival, audio analysis, or legacy Unix/Java workflows.
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
MTS files (AVCHD) store video as H.264 and audio as either AC-3 (Dolby Digital) or AAC, wrapped in an MPEG-2 Transport Stream container. During this conversion, FFmpeg discards the H.264 video stream entirely and decodes the compressed AC-3 or AAC audio track to raw PCM samples. Those samples are then re-encoded as 16-bit signed big-endian PCM and written into the AU container format. Unlike a simple remux, this process involves full audio decoding and re-encoding — the compressed camcorder audio is fully decompressed into uncompressed PCM. The result is a lossless representation of the audio content (no further compression artifacts are introduced beyond what was already in the original AC-3/AAC encode), stored in Sun's straightforward AU format with a minimal 28-byte header.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool. In the browser version of this tool, this runs via FFmpeg.wasm, a WebAssembly port of FFmpeg that executes entirely in your browser without any server upload. |
-i input.mts
|
Specifies the input AVCHD file in MTS format — the MPEG-2 Transport Stream container used by Sony and Panasonic camcorders, containing H.264 video and AC-3 or AAC audio streams. |
-vn
|
Disables video output entirely, instructing FFmpeg to ignore the H.264 video stream from the MTS file and write only the audio to the AU output. Without this flag, FFmpeg would attempt to include video, which the AU format cannot contain. |
-c:a pcm_s16be
|
Decodes the compressed AC-3 or AAC audio from the MTS file and re-encodes it as 16-bit signed big-endian PCM, which is the native and most widely compatible audio encoding for the Sun AU format. |
output.au
|
Specifies the output file in Sun AU format. FFmpeg infers the AU container from the .au file extension and writes the PCM audio data with a standard AU header containing the sample rate and channel count from the original MTS recording. |
Common Use Cases
- Extracting clean dialogue or ambient audio from Sony or Panasonic camcorder footage for use in audio editing software on Unix/Linux systems that natively support the AU format
- Archiving the audio track from AVCHD recordings in an uncompressed PCM format to avoid any further generation loss from re-encoding to another lossy codec
- Feeding camcorder audio into Java applications or legacy Sun/Unix audio pipelines that expect AU files with big-endian PCM encoding
- Separating audio from MTS video files captured during events or interviews to transcribe speech or perform acoustic analysis using Unix command-line tools
- Creating uncompressed AU reference files from AVCHD footage to use as a baseline when comparing audio quality across different codec conversion chains
- Preparing camcorder audio for import into MATLAB or scientific audio processing tools that support the AU format as a straightforward PCM container
Frequently Asked Questions
The conversion from MTS to AU involves decoding the compressed AC-3 or AAC audio in your AVCHD file to raw PCM, which does not introduce any new compression artifacts. However, because AC-3 and AAC are lossy codecs, the original camcorder recording already discarded some audio information at capture time — that loss cannot be recovered. The AU output will be an exact, uncompressed representation of the decoded audio, so no additional quality is lost in this conversion step.
The AU format stores audio as uncompressed PCM — every sample is stored as a raw 16-bit integer with no compression whatsoever. Your MTS file's audio was stored as AC-3 or AAC, which typically compresses audio to around 128–256 kbps. Uncompressed 16-bit stereo PCM at 48 kHz (a common camcorder sample rate) requires approximately 5.5 MB per minute, compared to under 2 MB per minute for compressed AC-3. This size difference is expected and is the cost of lossless uncompressed storage.
The AU format technically supports multiple channels, and FFmpeg will pass through the channel layout from your decoded MTS audio. However, the AU format has limited metadata capabilities compared to modern containers, and some applications may not correctly interpret multi-channel AU files. If your AVCHD footage was recorded in 5.1 surround, you may want to verify that your target application handles multi-channel AU correctly, or consider down-mixing to stereo before or after this conversion.
The command uses '-c:a pcm_s16be' to produce 16-bit signed big-endian PCM, which is the standard codec for AU files. The AU format also supports pcm_s8, pcm_u8, pcm_alaw, and pcm_mulaw. For example, to encode with G.711 mu-law (common in telephony), you would change the command to: ffmpeg -i input.mts -vn -c:a pcm_mulaw output.au. Note that these alternative codecs reduce bit depth or apply companding, so pcm_s16be generally offers the best fidelity.
Yes. On Linux or macOS, you can loop over all MTS files in a directory with: for f in *.mts; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.mts}.au"; done. On Windows Command Prompt, use: for %f in (*.mts) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.au". This is especially useful for bulk-processing footage from a camcorder memory card, and the browser-based tool is best suited for single files under 1GB.
AVCHD files can carry multiple audio tracks, but by default FFmpeg selects only the first audio stream for output. Since the AU format does not support multiple audio tracks, only one track can be written to the output file. If you need a specific audio track — for example, a secondary microphone feed — you can add '-map 0:a:1' to the command to select the second audio track: ffmpeg -i input.mts -vn -map 0:a:1 -c:a pcm_s16be output.au.
Technical Notes
The Sun AU format uses a simple fixed header (annotation, data offset, data size, encoding type, sample rate, and channel count) followed by raw PCM sample data — there is no complex metadata block, no ID3 tags, and no chapter or subtitle support. When extracting audio from MTS, FFmpeg fully decodes the MPEG-2 Transport Stream's AC-3 or AAC audio payload before re-encoding to pcm_s16be; this means the process is CPU-bound rather than a fast demux/remux. The big-endian byte order of pcm_s16be is a defining characteristic of the AU format (inherited from Sun SPARC architecture) and differs from the little-endian PCM used in WAV files — applications that read raw PCM must account for this. The original MTS sample rate (commonly 48000 Hz for AVCHD) is preserved in the AU header. Metadata such as recording timestamps, GPS data, or camcorder model information stored in the MTS container will not be carried over to the AU file. If the source MTS uses AC-3 audio (common on Panasonic AVCHD cameras), FFmpeg requires the AC-3 decoder, which is included in standard FFmpeg and FFmpeg.wasm builds.