MyMediaTools

Extract Audio from MTS to WAV — Free Online Tool

Extract uncompressed audio from AVCHD camcorder footage (.mts files) and save it as a WAV file. This tool strips the AC-3 or AAC audio track from the MPEG-2 Transport Stream container and re-encodes it to 16-bit PCM WAV — delivering broadcast-ready, lossless-quality audio that works in virtually every DAW, video editor, and audio tool.

FFmpeg Command

Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg

Free — no uploads, no signups. Your files never leave your browser.

Estimated output:

Conversion Complete!

Download

How It Works

MTS files store audio as either AC-3 (Dolby Digital) or AAC inside an MPEG-2 Transport Stream container alongside H.264 video. During this conversion, the video stream is discarded entirely using the -vn flag — no video decoding or encoding takes place. The compressed audio stream (AC-3 or AAC) is then decoded to raw PCM samples and written into a WAV container using the pcm_s16le codec, which stores audio as signed 16-bit little-endian integers at the source file's original sample rate and channel count. The result is a fully uncompressed WAV file — there is one lossy step (decoding the original compressed audio), but no further lossy compression is applied, so the WAV represents the highest-quality version of the audio that was captured by the camcorder.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg command-line tool. In this browser-based tool, the same FFmpeg engine runs locally in your browser via WebAssembly (FFmpeg.wasm) — no files leave your device.
-i input.mts Specifies the input file — an MTS file from an AVCHD camcorder such as a Sony Handycam or Panasonic HC series. FFmpeg reads the MPEG-2 Transport Stream container and identifies the H.264 video and AC-3 or AAC audio streams inside.
-vn Disables video output entirely. Since the goal is audio extraction, this flag tells FFmpeg to skip all video stream processing — the H.264 video track in the MTS file is not decoded, not encoded, and not included in the output WAV file, which makes the conversion significantly faster.
-c:a pcm_s16le Sets the output audio codec to PCM signed 16-bit little-endian, which is the standard uncompressed audio encoding used in WAV files. FFmpeg decodes the compressed AC-3 or AAC audio from the MTS source and re-encodes it as raw PCM samples, producing an uncompressed WAV that preserves the full audio quality of the original camcorder recording.
output.wav Specifies the output filename with the .wav extension. FFmpeg uses this extension to select the WAV container format automatically, which is the correct container for pcm_s16le audio and is compatible with virtually every DAW, video editor, audio editor, and broadcast tool available.

Common Use Cases

  • Pulling the audio from AVCHD wedding or event footage to hand off to an audio engineer for mixing or noise reduction before final video edit
  • Extracting interview audio recorded on a Sony or Panasonic camcorder to import into a DAW like Pro Tools, Logic Pro, or Adobe Audition for professional post-production
  • Converting camcorder audio to WAV for use as a clean reference track when syncing dual-system sound with a separate field recorder like a Zoom H6
  • Archiving the audio from irreplaceable MTS camcorder footage in an uncompressed format so it is not subject to further lossy re-encoding in future workflows
  • Extracting voice-over or narration recorded directly to a camcorder on set to use as guide audio in a video editing timeline
  • Preparing camcorder audio for transcription services or AI speech-to-text tools that require or prefer uncompressed WAV input

Frequently Asked Questions

There is one unavoidable quality step: the AC-3 or AAC audio in the MTS file is a lossy-compressed format, so the original camcorder recording was already lossy at the point of capture. Converting to WAV decodes that compressed audio to raw PCM, which is technically a generation loss, but at typical camcorder bitrates (192–384 kbps AC-3) this is audibly transparent. Once in WAV format, no further lossy compression is applied, so you are preserving the best-possible version of what the camcorder recorded.
By default, FFmpeg selects the first (default) audio stream in the MTS file, which is usually the primary stereo or 5.1 mix. Many Sony and Panasonic AVCHD camcorders record a single stereo AC-3 track, but some professional models record multiple tracks. If you need to extract a specific track, you can modify the command with -map 0:a:1 (for the second audio track) before the output filename when running it locally on your desktop.
MTS files store audio in a compressed format like AC-3 or AAC, which can achieve compression ratios of 10:1 or higher compared to uncompressed audio. WAV with pcm_s16le stores every audio sample as a raw 16-bit integer with no compression, so a stereo 48 kHz stream consumes roughly 5.6 MB per minute per channel. A 10-minute MTS file with AC-3 audio at 192 kbps might produce a WAV file 15–20 times larger — this is expected and correct behavior for an uncompressed format.
The WAV output will use the same sample rate as the audio in the source MTS file. AVCHD camcorders typically record at 48 kHz, which is the broadcast standard for video production audio. FFmpeg preserves the original sample rate during the decode-to-PCM process, so you do not need to specify it explicitly in the command. If you need to resample to 44.1 kHz for music production contexts, you can add -ar 44100 to the command before the output filename.
Yes. On Linux or macOS, you can run a shell loop: for f in *.mts; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.mts}.wav"; done. On Windows Command Prompt, use: for %f in (*.mts) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav". This is particularly useful for bulk-extracting audio from an entire session of camcorder clips, and since the video is discarded early in the process, conversion is very fast — typically much faster than real-time.
AVCHD camcorders record audio at 16-bit depth, so pcm_s16le is the natural and correct choice — you would not gain any real information by outputting to pcm_s24le or pcm_s32le, since the source material contains no audio data beyond 16 bits. If you want to future-proof the file for processing in a 32-bit float DAW environment, you can change pcm_s16le to pcm_f32le in the command, which stores the same audio data with headroom for internal processing, but the actual captured dynamic range remains bounded by the original 16-bit recording.

Technical Notes

AVCHD MTS files carry audio in one of two codecs depending on the camcorder model and recording mode: AC-3 (Dolby Digital), common on older Sony and Panasonic consumer and prosumer camcorders, or AAC-LC, used in some newer AVCHD implementations. Both are lossy codecs, and FFmpeg's libavcodec handles decoding of both transparently without requiring any special flags — the -c:a pcm_s16le instruction alone is sufficient to tell FFmpeg to decode whatever compressed audio it finds and write it as 16-bit PCM. The WAV container itself does not support metadata fields like artist or album in any standardized way, so camcorder-specific metadata embedded in the MTS (such as recording date or GPS coordinates) will not be preserved in the output — if metadata preservation matters, consider using FLAC or writing the metadata to a sidecar file. WAV files do not support chapters, multiple audio tracks, or subtitle streams. If the source MTS contains 5.1 surround audio, it will be extracted to a multi-channel WAV file correctly, though some consumer software may not correctly display or play back 6-channel WAV files. For maximum compatibility with audio software, stereo 48 kHz pcm_s16le WAV is universally supported across all major DAWs, NLEs, and broadcast playout systems.

Related Tools

Extract Audio MTS to AIFF Extract Audio MTS to FLAC Extract Audio MTS to ALAC Extract Audio 3G2 to WAV Extract Audio DV to WAV Extract Audio MPG to WAV Convert MTS to MP4 Compress MTS