Extract Audio from M2TS to WAV — Free Online Tool
Extract audio from M2TS Blu-ray and AVCHD files and save it as an uncompressed PCM WAV file. This tool strips the video stream entirely and decodes whatever audio codec is present in the M2TS container — whether that's Dolby TrueHD, DTS-HD, AAC, or AC-3 — into raw 16-bit PCM, giving you a universally compatible, uncompressed audio file.
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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 high-definition video alongside one or more audio tracks encoded in formats like Dolby TrueHD, DTS-HD Master Audio, AC-3 (Dolby Digital), or AAC. This tool discards the video stream entirely and decodes the primary audio track into uncompressed PCM audio, then wraps it in a WAV container. Unlike a remux, this is a full decode-and-re-encode step: the compressed audio bitstream from the M2TS is decoded to raw PCM samples, then written as 16-bit little-endian PCM (pcm_s16le) into the WAV file. If the source audio was lossless (e.g., TrueHD or DTS-HD MA), the resulting WAV is a true lossless representation of that audio at 16-bit depth. If the source was lossy (e.g., AC-3 or AAC), the WAV preserves the audio exactly as decoded — no additional quality loss occurs in this step, but the original lossy compression artifacts remain.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg media processing tool. All subsequent flags and arguments define the input source, processing steps, and output destination for this M2TS-to-WAV extraction. |
-i input.m2ts
|
Specifies the input M2TS file — an MPEG-2 Transport Stream container as used on Blu-ray discs and AVCHD camcorders. FFmpeg reads and demuxes all streams (video, audio, subtitles) from this file, making them available for the processing pipeline. |
-vn
|
Disables all video output, telling FFmpeg to ignore and discard the video stream from the M2TS entirely. This is essential for audio extraction — without it, FFmpeg would attempt to include video in the output, which WAV cannot contain. |
-c:a pcm_s16le
|
Decodes the M2TS audio track (which may be AC-3, TrueHD, DTS, or AAC) and re-encodes it as 16-bit signed PCM in little-endian byte order — the standard uncompressed audio format for WAV files and the most universally compatible PCM variant across Windows, macOS, and Linux applications. |
output.wav
|
Defines the output file as a WAV container. FFmpeg infers the WAV format from the .wav extension and writes the uncompressed PCM audio stream into it, producing a file directly usable in any DAW, audio editor, or media player without further decoding. |
Common Use Cases
- Extracting the lossless audio track from a Blu-ray rip (.m2ts) to archive or master-quality audio before re-encoding to a smaller format like FLAC or AAC
- Pulling dialogue or sound design from AVCHD camcorder footage (.m2ts) into a DAW like Pro Tools or Reaper, which require uncompressed PCM WAV for editing
- Separating the audio from a Blu-ray concert film or live recording to burn to an audio CD or distribute as a high-quality WAV download
- Extracting broadcast-captured M2TS content for use in video editors that handle WAV audio more reliably than transport stream audio streams
- Converting the AC-3 or AAC audio from an M2TS file to a format compatible with Windows legacy applications that do not support compressed audio containers
- Generating a WAV reference file from a Blu-ray source to use as a synchronization reference when mixing or conforming audio in post-production
Frequently Asked Questions
If your M2TS file contains a lossless audio track such as Dolby TrueHD or DTS-HD Master Audio, converting to WAV with pcm_s16le introduces no perceptible quality loss — you get a full, uncompressed representation of that audio. If the source track is lossy (AC-3, AAC, or standard DTS), decoding it to PCM WAV does not degrade it further, but those original compression artifacts from the M2TS are already baked in. WAV itself adds no additional compression, so no new quality is lost in this conversion step.
FFmpeg selects the first audio track by default, which is typically the primary language track or the highest-priority stream as declared in the M2TS container. Blu-ray M2TS files often carry multiple tracks — for example, a Dolby TrueHD track alongside an AC-3 compatibility track. If you want to extract a specific track, you can modify the FFmpeg command to add '-map 0:a:1' (for the second audio stream, zero-indexed) before the output filename. WAV does not support multiple audio tracks, so only one stream can be written per output file.
M2TS files store audio in compressed formats like AC-3 or TrueHD, which dramatically reduce file size. WAV with pcm_s16le stores every audio sample as raw, uncompressed data — a 5.1 surround track at 48 kHz, 16-bit takes roughly 4.6 MB per minute per channel, meaning a two-hour film's audio alone could exceed 3 GB. This is expected behavior: WAV prioritizes universal compatibility and zero-loss representation over storage efficiency. If file size is a concern, consider converting to FLAC instead, which is losslessly compressed and far smaller.
Yes, WAV supports multi-channel PCM audio including 5.1 and 7.1 surround configurations, and FFmpeg will preserve the channel layout from the M2TS source. However, some older applications and devices that accept WAV files may only handle stereo (2-channel) audio correctly. If your downstream tool struggles with the surround WAV, you can add '-ac 2' to the FFmpeg command to downmix to stereo during the conversion.
You can add time-based flags to the FFmpeg command to extract only a portion of the audio. Insert '-ss 00:05:00' before the input to seek to the five-minute mark, and '-t 00:10:00' after the input to limit output to ten minutes. For example: 'ffmpeg -ss 00:05:00 -i input.m2ts -t 00:10:00 -vn -c:a pcm_s16le output.wav'. Placing '-ss' before '-i' uses fast keyframe seeking, which is efficient for large M2TS files but may be slightly imprecise; placing it after '-i' is frame-accurate but slower.
Yes. On Linux or macOS, you can loop over files in a directory with a shell command: 'for f in *.m2ts; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.m2ts}.wav"; done'. On Windows Command Prompt, use: 'for %f in (*.m2ts) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav"'. Each M2TS file is processed sequentially, producing a matching WAV file with the same base name. This is especially useful when ripping a full Blu-ray disc that stores each title or episode as a separate .m2ts file.
Technical Notes
The pcm_s16le codec used in this conversion writes 16-bit signed integer samples in little-endian byte order, which is the WAV standard and compatible with virtually every audio application, DAW, and operating system. M2TS files from Blu-ray sources often carry audio at 24-bit depth (particularly TrueHD and DTS-HD MA streams), so this conversion involves bit-depth reduction from 24-bit to 16-bit, which technically discards the lower 8 bits of precision. For most listening purposes this is inaudible, but mastering engineers who need to preserve the full 24-bit depth should modify the command to use '-c:a pcm_s24le' and save to a 24-bit WAV. WAV does not support metadata fields like track title, artist, or chapter markers, so any metadata embedded in the M2TS container is not carried over. Additionally, WAV has a theoretical 4 GB file size limit due to its 32-bit chunk size header — very long, high-channel-count audio extracted from M2TS could approach this limit, in which case the RF64 extension (a WAV superset) would be needed. The sample rate of the source audio (typically 48 kHz for broadcast/Blu-ray content, occasionally 96 kHz for high-resolution titles) is preserved as-is in the output WAV without resampling.