Convert MTS to AIF — Free Online Tool
Extract and convert the audio track from an MTS camcorder video into a lossless AIF file, decoding the compressed AC-3 or AAC audio from the AVCHD stream into full 16-bit PCM for uncompressed playback on Mac systems.
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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 store audio in a lossy compressed format — either AC-3 (Dolby Digital) or AAC — wrapped inside an MPEG-2 Transport Stream container alongside an H.264 video track. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio to raw PCM samples, which are then written into an AIF container as 16-bit big-endian PCM (pcm_s16be). This is a full decode-and-re-encode of the audio stage: the compressed codec data is fully decompressed to lossless PCM. The result is larger than the original audio portion of the MTS file, but is now uncompressed and ready for professional audio editing or archival on Mac.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In this browser-based tool, the same command runs via FFmpeg.wasm compiled to WebAssembly, so the behavior and output are identical to running it locally on your desktop. |
-i input.mts
|
Specifies the input file — an MTS file recorded by an AVCHD camcorder such as a Sony Handycam or Panasonic HDC series. FFmpeg reads the MPEG-2 Transport Stream container and demuxes the H.264 video and compressed audio streams from it. |
-c:a pcm_s16be
|
Sets the audio codec to 16-bit signed big-endian PCM, which is the standard uncompressed format for AIF files. This fully decodes the lossy AC-3 or AAC audio from the MTS stream into raw uncompressed PCM samples compatible with the AIF container spec and Mac audio tools. |
output.aif
|
Defines the output filename and tells FFmpeg to write an AIF container. FFmpeg infers the AIF format from the .aif extension and packages the decoded PCM audio into the correct big-endian AIF chunk structure expected by macOS applications like Logic Pro, GarageBand, and QuickTime. |
Common Use Cases
- Extracting a clean, uncompressed audio track from a Sony or Panasonic camcorder recording for import into Logic Pro or GarageBand on Mac
- Archiving the spoken-word or ambient audio from an AVCHD camcorder clip as a lossless AIF file before the original footage is edited or deleted
- Pulling interview audio recorded on a DSLR or consumer camcorder in MTS format for use in a podcast production workflow where lossless source files are required
- Converting camcorder event footage audio to AIF so it can be mixed and processed in a Mac-based DAW without introducing generation loss from re-encoding lossy formats
- Recovering and preserving the audio layer from corrupted or partially-readable MTS files where only the audio stream can be decoded successfully
Frequently Asked Questions
There is a one-time decode loss, because the original audio in the MTS file is already compressed — either as AC-3 or AAC, both of which are lossy formats. FFmpeg fully decodes that compressed audio to raw PCM and writes it into the AIF container without any further compression. The resulting AIF is a lossless snapshot of what the decoder produced, so no additional quality is lost beyond what was already lost when the camcorder originally encoded the audio.
The MTS file stores audio in a compressed format like AC-3 or AAC, which achieves significant size reductions through lossy compression. AIF stores audio as uncompressed PCM, where every sample is written out in full. A 16-bit stereo 48kHz PCM stream — typical for camcorder audio — uses roughly 5.6 MB per minute per channel, compared to AC-3 at around 192–384 kbps. Expect the extracted AIF to be several times larger than the compressed audio it came from.
Yes. FFmpeg preserves the original sample rate from the MTS audio stream when writing the AIF output. Camcorder recordings in AVCHD format typically use a 48 kHz sample rate, which is the broadcast standard, and this will be carried through to the AIF file. No resampling occurs unless you explicitly add a -ar flag to the command.
Yes. The default command uses pcm_s16be, which produces a 16-bit AIF file. To get 24-bit output, change the audio codec flag to -c:a pcm_s24be, which writes 24-bit big-endian PCM — the format commonly used for professional audio work on Mac. You can also use pcm_s32be for 32-bit integer or pcm_f32be for 32-bit float, depending on your DAW's requirements.
The H.264 video stream in the MTS file is completely dropped. FFmpeg automatically selects only the audio stream when writing to an AIF output, since AIF is a pure audio container with no video support. You do not need to add a -vn flag explicitly, though doing so makes the intent explicit if you are adapting the command for scripting purposes.
On macOS or Linux, you can use a shell loop: for f in *.mts; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.mts}.aif"; done. This iterates over every MTS file in the current directory and produces a matching AIF file with the same base name. On Windows Command Prompt, use: for %f in (*.mts) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aif". This is especially useful for processing multi-clip camcorder sessions recorded as sequential MTS files.
Technical Notes
AVCHD MTS files record audio at 48 kHz, the standard broadcast sample rate, in either AC-3 (Dolby Digital, common on older Sony and Panasonic camcorders) or AAC stereo. When decoding to pcm_s16be for AIF, FFmpeg uses its internal AC-3 or AAC decoder and writes big-endian 16-bit samples, which is the native byte order for the AIF format (AIF is a big-endian format by spec, in contrast to its little-endian sibling AIFF-C). The AIF container does not support multiple audio tracks, subtitles, or chapters, so only the first (or default) audio stream from the MTS file will be written. If your MTS file contains multiple audio tracks — for example, a dual-channel stereo and a surround mix — only one will be extracted unless you specify -map explicitly in the command. Metadata such as track title or recording timestamp embedded in the AVCHD transport stream is generally not preserved in the AIF output, as the AIF chunk-based metadata structure differs significantly from MPEG-2 TS metadata. For professional audio archival, 24-bit output via pcm_s24be is recommended if the destination DAW supports it, since it provides headroom that can be valuable during post-processing even when the source was 16-bit.