Extract Audio from MTS to WEBA — Free Online Tool
Extract audio from AVCHD camcorder footage (.mts files) and save it as a WEBA file encoded with the Opus codec — ideal for web-optimized audio that's smaller and higher quality than MP3 at equivalent bitrates. This tool strips the H.264 video and AC-3/AAC audio from the MPEG-2 Transport Stream container and re-encodes just the audio into the WebM-based WEBA format.
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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
MTS files from Sony and Panasonic camcorders use the AVCHD standard: an MPEG-2 Transport Stream container holding H.264 video alongside AC-3 or AAC audio. During this conversion, the video stream is discarded entirely using the -vn flag — no video decoding or encoding happens. The existing audio stream (AC-3 or AAC) is then decoded and re-encoded using the Opus codec (libopus) inside a WebM container with the .weba extension. Opus is a modern, royalty-free codec that delivers excellent quality at low bitrates, making WEBA files significantly more web-friendly than the original broadcast-oriented audio from camcorder footage. Because the source AC-3 or AAC audio must be decoded and re-encoded into Opus, this is a full transcoding operation — not a lossless copy — so some generation loss applies.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary. In this browser-based tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm), so your MTS camcorder file never leaves your device. |
-i input.mts
|
Specifies the input AVCHD file in MPEG-2 Transport Stream format. FFmpeg reads the MTS container and identifies both the H.264 video stream and the AC-3 or AAC audio stream inside. |
-vn
|
Disables video output entirely, instructing FFmpeg to ignore the H.264 video stream from the AVCHD footage. This is what makes the operation an audio extraction rather than a video conversion, and it significantly speeds up processing since no video decoding or encoding occurs. Note: this flag appears twice in the resolved command — once before and once after the audio codec flags — which is harmless redundancy; a single -vn is sufficient. |
-c:a libopus
|
Selects the Opus encoder (libopus) to re-encode the audio extracted from the MTS file. Opus is required for WEBA output and is superior to the original AC-3 or AAC audio at web-targeted bitrates, offering better quality-per-bit for camcorder audio content. |
-b:a 128k
|
Sets the Opus audio bitrate to 128 kilobits per second. For typical camcorder audio (speech, ambient sound, event audio), 128k Opus provides transparent or near-transparent quality. Lower values like 64k or 96k are viable for voice-only content; raise to 192k or 256k for music or high-fidelity recordings. |
output.weba
|
Defines the output filename with the .weba extension, which signals FFmpeg to wrap the Opus audio stream in a WebM container. The WEBA format is the audio-only variant of WebM and is designed for direct embedding and playback in modern web browsers. |
Common Use Cases
- Pull the spoken audio from a camcorder interview recording to create a podcast episode or voiceover track without carrying the large AVCHD video file
- Extract ambient or event audio captured on a Sony or Panasonic camcorder to embed directly in a web page using the HTML5 <audio> element, which natively supports WEBA/Opus
- Strip the audio from wedding or event camcorder footage to archive just the speeches or ceremony audio at a fraction of the original file size
- Prepare audio commentary or narration recorded on a camcorder for use in a web-based video editor or content management system that expects WebM-compatible audio
- Extract the AC-3 surround audio track from AVCHD footage and convert it to a stereo Opus stream for use in browser-based applications where AC-3 is not supported
- Quickly isolate the audio layer from MTS clips shot on location to review audio quality before committing to a full video editing workflow
Frequently Asked Questions
Yes, some quality loss occurs because this is a full transcode. The original AC-3 or AAC audio in the MTS file must be fully decoded and then re-encoded using the Opus codec. That said, Opus is exceptionally efficient — at the default 128k bitrate, Opus typically sounds equal to or better than AAC or MP3 at the same bitrate. If your source MTS was recorded with AC-3 at 192k or higher, setting the output bitrate to 192k (-b:a 192k) will better preserve the original quality.
By default, FFmpeg will pass all audio channels through the Opus encoder, which does support multichannel audio including 5.1 surround. However, WEBA files with multichannel Opus audio have limited browser playback support, and many web players expect stereo. If you need stereo output, you can add '-ac 2' to the FFmpeg command to downmix the surround channels to stereo before encoding. For archiving the full surround track, the default behavior preserves the channel layout.
WEBA is a WebM container with Opus audio, and it is natively supported in Chrome, Firefox, and Edge. Safari added Opus support starting with Safari 17 (macOS Sonoma and iOS 17), so most modern browsers now handle it. Android's native media player generally supports Opus/WebM as well. Where WEBA lacks support — for example, older Apple devices or some embedded players — you may need to convert to MP3 or AAC/M4A instead.
Replace the '128k' value in the '-b:a 128k' portion of the command with your desired bitrate. For speech-only content from camcorder recordings (interviews, event narration), 64k or 96k Opus delivers very intelligible audio at a tiny file size. For music or high-fidelity ambient audio captured by the camcorder, 192k or 256k will better preserve detail. The full adjusted command would look like: ffmpeg -i input.mts -vn -c:a libopus -b:a 192k -vn output.weba
Yes. On Linux or macOS, you can run: for f in *.mts; do ffmpeg -i "$f" -vn -c:a libopus -b:a 128k -vn "${f%.mts}.weba"; done — this loops through every MTS file in the current directory and extracts the audio to a matching WEBA file. On Windows PowerShell, use: Get-ChildItem *.mts | ForEach-Object { ffmpeg -i $_.FullName -vn -c:a libopus -b:a 128k -vn ($_.BaseName + '.weba') }. This is especially useful when dumping an entire camcorder memory card.
AVCHD MTS files store recording metadata (date, camera model, GPS coordinates) in a proprietary sidecar structure or within MPEG-2 Transport Stream private data sections that FFmpeg does not automatically map to WebM metadata tags. Standard ID3-style tags like title or artist are not typically embedded in raw MTS files either. If you want to add metadata to the WEBA output, you can append flags like -metadata title="Interview 2024" -metadata artist="Camera Operator" to the FFmpeg command before the output filename.
Technical Notes
MTS files use the MPEG-2 Transport Stream (M2TS) container, which is optimized for broadcast and camcorder recording rather than web delivery. The audio inside is typically AC-3 (Dolby Digital) at 192–256 kbps or AAC at 128–256 kbps, and this audio must be fully decoded before being re-encoded into Opus for the WEBA output — there is no stream-copy path between AC-3/AAC and Opus. The WEBA format (audio-only WebM) supports only Opus and Vorbis codecs; this tool uses Opus (libopus) as it consistently outperforms Vorbis at all bitrates and is the modern standard for WebM audio. One known limitation: WEBA does not support multiple audio tracks, so if your MTS file contains more than one audio stream (e.g., a main mix and a camera microphone track), only the first audio stream will be extracted by default. To select a different track, add '-map 0:a:1' to the command. Chapters and subtitle data from the MTS file are not preserved in the WEBA output, as the format supports neither. File sizes will typically be 80–95% smaller than the original MTS file since the video stream (which dominates MTS file size) is discarded entirely.