Convert M2TS to WAV — Free Online Tool
Extract uncompressed PCM audio from M2TS Blu-ray and AVCHD files, converting the audio stream to a 16-bit WAV file using the pcm_s16le codec. Ideal for archiving, editing, or analyzing high-definition video audio without any lossy compression.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your M2TS file here
or click to browse
Free — no uploads, no signups. Your files never leave your browser.
Settings
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 video (H.264/AVC or H.265/HEVC) alongside audio encoded in formats like AAC, AC-3, DTS, or TrueHD — often at high bitrates for Blu-ray quality. This conversion discards the video stream entirely and decodes whichever audio track is present in the M2TS container, then re-encodes it as raw 16-bit signed little-endian PCM audio wrapped in a WAV container. Because the source audio codec (e.g., AAC or AC-3) is lossy, this conversion involves a decode step — the compressed audio is fully decoded to raw PCM samples. The resulting WAV is uncompressed and lossless from that decoded point forward, meaning no further quality is lost after extraction, though any quality loss from the original M2TS audio encoding is already baked in.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, which handles all demuxing, decoding, and encoding. In the browser version this runs via FFmpeg.wasm (WebAssembly), executing the same logic locally in your browser without uploading your M2TS file to any server. |
-i input.m2ts
|
Specifies the input file — an M2TS container holding an MPEG-2 Transport Stream, typically from a Blu-ray disc rip or AVCHD camcorder. FFmpeg will demux this to access the video, audio, and any subtitle streams inside. |
-c:a pcm_s16le
|
Sets the audio codec for the output to 16-bit signed little-endian PCM — the standard uncompressed audio format used in WAV files. The M2TS audio (e.g., AC-3, AAC) is fully decoded and re-encoded as raw PCM samples, producing an uncompressed WAV with no further quality loss from this point forward. |
output.wav
|
Defines the output filename and container format. The .wav extension tells FFmpeg to wrap the PCM audio in a RIFF/WAV container, which is the standard uncompressed audio file format developed by Microsoft and IBM, widely supported across all platforms and audio software. |
Common Use Cases
- Extracting dialogue or soundtrack audio from Blu-ray rips for use in a professional Digital Audio Workstation (DAW) like Pro Tools or Reaper, which expects standard PCM WAV input
- Pulling audio from AVCHD camcorder footage (which uses the .m2ts container) before importing into video editing software that struggles with AVCHD audio streams
- Creating a broadcast-ready WAV master from a Blu-ray source file for dubbing, subtitling, or localization workflows that require uncompressed audio
- Archiving the audio track of a home video recorded to Blu-ray disc in an uncompressed, widely compatible format that will remain readable for decades without codec dependencies
- Analyzing audio levels, waveforms, or audio sync issues in an M2TS file using audio tools that require uncompressed PCM input rather than compressed transport stream audio
- Separating the audio from a multi-hour M2TS concert or event recording to produce standalone audio files for distribution or editing
Frequently Asked Questions
Not fully — it depends on what's inside the M2TS. If the M2TS carries a lossy audio codec like AC-3 (Dolby Digital) or AAC, the audio was already compressed with quality loss when the M2TS was created. The conversion decodes that lossy audio to raw PCM and stores it in WAV without any further loss, but the original lossy compression artifacts are preserved. If your M2TS contains a lossless track such as TrueHD or DTS-HD MA (common on some Blu-ray rips), the decoded WAV will represent that audio at full fidelity.
The M2TS audio stream is compressed — AC-3 at 640kbps or AAC at 192–256kbps takes up far less space than uncompressed PCM. When converted to 16-bit PCM WAV at standard sampling rates (e.g., 48kHz stereo), the audio requires roughly 5–10 MB per minute for stereo or significantly more for 5.1 surround. A two-hour Blu-ray rip's audio track could expand from around 500MB compressed to several gigabytes as uncompressed WAV, depending on channel count.
No — by default, FFmpeg selects only the best single audio stream (usually the first or highest-quality track) from the M2TS container. M2TS files from Blu-ray discs often contain multiple audio tracks for different languages or formats. To extract a specific track, you would add a stream selector like -map 0:a:1 to target the second audio stream. The WAV format also does not support multiple audio tracks in a single file, so each track would need to be extracted separately.
FFmpeg will preserve the channel layout from the source audio during decoding. If the M2TS contains a 5.1 AC-3 or DTS track, the resulting WAV will contain 6 channels of PCM audio (front left, front right, center, LFE, surround left, surround right) encoded as pcm_s16le. Most consumer playback software handles multi-channel WAV files, but some tools may only read stereo WAV. If you need stereo output, you can add -ac 2 to the FFmpeg command to downmix to stereo.
Replace -c:a pcm_s16le with -c:a pcm_s24le in the command to produce a 24-bit signed little-endian PCM WAV file. This is particularly useful if your M2TS source contains a high-fidelity lossless audio track (such as TrueHD) and you want to preserve full dynamic range for professional audio work. The command would then read: ffmpeg -i input.m2ts -c:a pcm_s24le output.wav. Note that 24-bit WAV files will be approximately 50% larger than 16-bit equivalents.
Yes — on Linux or macOS you can use a shell loop: for f in *.m2ts; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.m2ts}.wav"; done. On Windows Command Prompt, use: for %f in (*.m2ts) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This is especially useful for processing multiple AVCHD clips from a camcorder memory card or a folder of Blu-ray chapter files.
Technical Notes
The pcm_s16le codec produces standard 16-bit signed little-endian PCM, the most universally supported WAV variant — compatible with virtually every audio editor, DAW, broadcast playout system, and operating system audio API. M2TS containers use MPEG-2 Transport Stream framing, which includes timing and sync information for broadcast use; all of this container-level metadata is stripped during conversion since WAV has no equivalent metadata structure. Beyond basic RIFF header fields, WAV does not support rich metadata like ID3 tags, so any title, artist, or chapter information embedded in the M2TS will be lost. The sample rate of the output WAV will match the source audio (typically 48kHz for Blu-ray and AVCHD sources, rather than the 44.1kHz common in music), which is correct for video production workflows. If you require 44.1kHz output for music distribution, add -ar 44100 to the command. M2TS files with HDMV PGS subtitles or multiple video angles will have those streams silently ignored during audio-only extraction.