Convert MTS to WAV — Free Online Tool
Extract clean, uncompressed audio from AVCHD camcorder footage (.mts files) and save it as a broadcast-ready WAV file. This tool strips the H.264 video stream and decodes the AC-3 or AAC audio track into uncompressed 16-bit PCM — the lossless standard used in professional audio production and video editing.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your MTS file here
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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
MTS files use the MPEG-2 Transport Stream container, typically carrying H.264 video alongside AC-3 or AAC audio as recorded by Sony and Panasonic camcorders. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio (whether AC-3 or AAC) into raw, uncompressed PCM samples. These samples are then written into a WAV container using the pcm_s16le codec — signed 16-bit little-endian PCM — which is the standard encoding used in CD-quality audio and accepted by virtually every digital audio workstation (DAW), video editor, and broadcast tool. Because MTS audio is lossy (AC-3 or AAC), the output WAV will be uncompressed but not higher quality than the source; you are 'freezing' the audio at its current quality into a lossless container.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that powers this conversion. In the browser version of this tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) — your MTS file never leaves your device. |
-i input.mts
|
Specifies the input file — your AVCHD camcorder footage in MTS format. FFmpeg probes the MPEG-2 Transport Stream to identify the video (H.264) and audio (AC-3 or AAC) streams it contains before processing begins. |
-c:a pcm_s16le
|
Sets the audio codec to pcm_s16le — signed 16-bit little-endian PCM — which decodes the compressed AC-3 or AAC audio from the MTS file into uncompressed audio samples. This is the standard encoding for CD-quality WAV audio and is universally compatible with DAWs, video editors, and broadcast tools. |
output.wav
|
Defines the output filename and format. The .wav extension tells FFmpeg to write a RIFF WAV container, which wraps the uncompressed pcm_s16le audio data. No video stream is written because WAV is an audio-only format — the H.264 video from the MTS file is automatically discarded. |
Common Use Cases
- Extracting dialogue or ambient sound recorded on a Sony or Panasonic camcorder to import into a DAW like Audacity, Adobe Audition, or Logic Pro for mixing and post-production
- Pulling the audio track from AVCHD event footage (weddings, conferences, live performances) to deliver a standalone audio file to a client without sharing the raw video
- Preparing camcorder audio for broadcast submission, where WAV PCM is a required deliverable format and compressed formats like AC-3 or AAC are not accepted
- Creating a clean reference track from MTS footage to sync with separately recorded audio from a dedicated field recorder during multi-camera editing
- Archiving the audio component of AVCHD recordings in an uncompressed, format-agnostic WAV file to ensure long-term compatibility regardless of codec support in future software
- Extracting interview or voiceover audio recorded in-camera on AVCHD gear for transcription services that require standard WAV input
Frequently Asked Questions
No. The audio in an MTS file is encoded as either AC-3 (Dolby Digital) or AAC — both lossy codecs — meaning some audio data was permanently discarded at the time of recording. Converting to WAV with pcm_s16le decodes that compressed audio into uncompressed PCM, but it cannot recover the information that was lost during the original encoding. The WAV will be a perfect, uncompressed representation of what was in the MTS, but it will not sound better than the source.
Most consumer AVCHD camcorders record stereo audio (2 channels), so the resulting WAV will typically be stereo. If your MTS file contains a multi-channel AC-3 track (e.g., 5.1 surround), FFmpeg will decode all channels and write them into the WAV file. However, note that WAV only supports a single audio track — if your MTS has multiple audio tracks, only the first track is extracted by default. You would need to run separate FFmpeg commands with the -map flag to extract additional tracks.
This is expected behavior. The audio in your MTS file was stored in compressed form (AC-3 typically at 192–384 kbps, AAC typically at 128–256 kbps), while WAV with pcm_s16le stores every audio sample uncompressed. CD-quality stereo PCM audio requires approximately 10 MB per minute. A 1-hour MTS recording might have a few hundred megabytes of compressed audio, which expands to roughly 600 MB as uncompressed WAV. The video is discarded entirely, so the WAV will only contain audio data.
Replace pcm_s16le in the FFmpeg command with your desired codec: use pcm_s24le for 24-bit audio (standard in professional production), pcm_s32le for 32-bit integer, or pcm_f32le for 32-bit float. The full command would be: ffmpeg -i input.mts -c:a pcm_s24le output.wav. Note that upsampling bit depth beyond the quality of the original AC-3 or AAC source does not add real audio information, though 24-bit output is often preferred as an interchange format in DAW workflows.
Yes, on the command line you can use a shell loop to process multiple files. On Linux or macOS, run: for f in *.mts; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.mts}.wav"; done. On Windows Command Prompt, use: for %f in (*.mts) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This processes each MTS file in the current directory and creates a matching WAV file. The browser-based tool on this page processes one file at a time, so the FFmpeg command is particularly useful for batch workflows.
MTS/AVCHD files embed metadata in the transport stream and accompanying AVCHD directory structure (BDMV folders), and WAV has limited metadata support compared to formats like FLAC or MP4. FFmpeg will attempt to map basic metadata tags (such as creation time) into the WAV INFO chunk, but camcorder-specific metadata like GPS coordinates, scene modes, or camera model information is unlikely to be preserved. If metadata retention is critical, consider keeping the original MTS file as your archive master.
Technical Notes
MTS is not a standalone container in the traditional sense — it is a segment of an AVCHD stream that relies on the broader AVCHD directory structure (with BDMV and STREAM folders) for full chapter and clip metadata. When FFmpeg reads a raw .mts file, it probes the transport stream directly and identifies the first audio PID (Packet Identifier) for extraction. Consumer camcorders using AVCHD predominantly record AC-3 stereo at 48 kHz and 16-bit depth, which maps cleanly to the default pcm_s16le WAV output (also 48 kHz, 16-bit). Some models use AAC instead of AC-3, particularly in AVCHD Lite implementations; FFmpeg handles both transparently. The output sample rate is inherited from the source — no resampling is performed unless explicitly requested with the -ar flag. WAV files use RIFF chunk headers and have a theoretical 4 GB file size limit (due to 32-bit chunk size fields), though most modern software handles the RF64 extension for larger files. For files over 4 GB, consider using a format like FLAC or AIFF as an alternative lossless output.