Extract Audio from MTS to AIF — Free Online Tool
Extract clean, uncompressed audio from AVCHD camcorder footage (.mts) and save it as a lossless AIF file. This tool decodes the AC-3 or AAC audio track embedded in the MPEG-2 Transport Stream and outputs it as 16-bit big-endian PCM, preserving full audio fidelity in a Mac-native format ideal for professional editing.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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 as lossy compressed streams — typically AC-3 (Dolby Digital) or AAC — muxed inside an MPEG-2 Transport Stream container alongside H.264 video. This conversion discards the video stream entirely and decodes the compressed audio to raw PCM samples, then writes them into an AIF container using the pcm_s16be codec (16-bit signed big-endian PCM). Because AIF is uncompressed, the output file will be significantly larger than the original audio stream — but this is the nature of lossless uncompressed audio. Importantly, the AC-3 or AAC audio is fully decoded before being written, so the AIF output represents the cleanest possible version of what the camcorder originally captured, free of any container or compression artifacts.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool. In the browser version, this runs via FFmpeg.wasm compiled to WebAssembly, executing entirely on your device without any server upload. |
-i input.mts
|
Specifies the input file — an MTS file from an AVCHD camcorder. FFmpeg reads the MPEG-2 Transport Stream container and identifies the H.264 video and AC-3 or AAC audio streams inside it. |
-vn
|
Disables video output entirely. Since we only want the audio track from the MTS footage, this flag tells FFmpeg to ignore the H.264 video stream and not include it in the output, keeping the AIF file audio-only. |
-c:a pcm_s16be
|
Sets the audio codec to 16-bit signed big-endian PCM, which is the standard uncompressed audio encoding for AIF files. FFmpeg decodes the source AC-3 or AAC audio and re-encodes it as raw PCM samples in the byte order that AIF requires. |
output.aif
|
Specifies the output filename with the .aif extension. FFmpeg uses this extension to select the AIF/AIFF container format, which wraps the pcm_s16be audio stream in the structure expected by macOS audio applications like Logic Pro, GarageBand, and QuickTime. |
Common Use Cases
- Extracting the stereo or surround audio recorded by a Sony or Panasonic AVCHD camcorder for import into Logic Pro or GarageBand, which prefer AIF files natively on macOS
- Pulling the audio track from wedding or event footage recorded in MTS format to send to a sound engineer for mixing or mastering in a DAW that requires uncompressed source material
- Archiving the audio portion of AVCHD camcorder recordings in a lossless, uncompressed format to ensure long-term compatibility without codec dependency
- Separating dialogue or ambient sound recorded on a camcorder during a film shoot so it can be synced and edited independently in Final Cut Pro or Premiere Pro
- Converting MTS audio tracks to AIF for use as sample material in music production, where uncompressed PCM source files are strongly preferred over lossy originals
- Extracting narration or voiceover recorded directly onto a camcorder in the field for cleanup and editing in an audio workstation without the overhead of decoding video
Frequently Asked Questions
There is a subtle but important nuance here. The original audio in an MTS file is already lossy — recorded by the camcorder as AC-3 or AAC. Converting to AIF doesn't recover quality that was lost during that initial compression, but it does introduce no additional degradation. The AIF output is an exact, lossless PCM representation of what the lossy codec decoded, so you won't lose anything further in the process. Think of it as freezing the quality at its current state in an uncompressed container.
The MTS file stores audio as compressed AC-3 or AAC data, which is very compact — typically 128–384 kbps. AIF stores audio as raw uncompressed PCM, where a stereo 16-bit stream at 48 kHz (common for camcorder audio) requires about 1.5 MB per second. A 10-minute MTS clip might have only 50–80 MB of compressed audio, but the equivalent AIF file could be 900 MB or more. This size increase is expected and is the defining characteristic of uncompressed audio formats.
Yes. FFmpeg preserves the channel layout from the source MTS audio stream. If the camcorder recorded stereo (2-channel) audio, the AIF output will also be stereo. If the MTS file contains a 5.1 AC-3 surround track, the PCM output will retain all six channels in the AIF file. The channel count is determined by what the camcorder recorded, not by this conversion.
Yes. The default FFmpeg command uses pcm_s16be for 16-bit output, but you can switch to pcm_s24be for 24-bit or pcm_s32be for 32-bit by changing the -c:a flag. For example: ffmpeg -i input.mts -vn -c:a pcm_s24be output.aif. Since the source audio was captured by a consumer or prosumer camcorder at 16-bit or 24-bit depth, using pcm_s24be is a reasonable choice if you want headroom for processing in a DAW without integer overflow.
On macOS or Linux, you can run a shell loop: for f in *.mts; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.mts}.aif"; done. On Windows Command Prompt: for %f in (*.mts) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aif". This processes each MTS file in the current directory and saves a corresponding AIF file with the same base name, which is especially useful when you have a full memory card of AVCHD footage to extract audio from.
MTS files often carry minimal metadata in their transport stream headers, and AIF has limited metadata support compared to formats like FLAC or MP4. FFmpeg will attempt to copy any available metadata tags to the AIF file, but camcorder-specific EXIF-style metadata (recording date, GPS, camera model) embedded in the AVCHD structure is typically not transferred. If preserving that metadata is important, consider keeping the original MTS file alongside the extracted AIF.
Technical Notes
MTS files produced by Sony and Panasonic AVCHD camcorders most commonly carry AC-3 (Dolby Digital) audio at 48 kHz, with stereo or 5.1 channel layouts depending on the recording mode. Some models use AAC instead. In both cases, FFmpeg's internal AC-3 and AAC decoders handle decoding transparently, so the -c:a pcm_s16be flag simply specifies the output codec without needing to identify which input codec was used. The AIF format uses big-endian byte ordering (a legacy of its Apple/Motorola heritage), which is why pcm_s16be is the correct codec rather than the little-endian pcm_s16le used in WAV files — AIF files written with little-endian PCM may appear to open but will play back as noise in many applications. The output sample rate will match the source (typically 48 kHz for camcorder footage), and no resampling occurs unless explicitly requested. One known limitation: if an MTS file contains multiple audio tracks (some AVCHD cameras record a second audio channel for backup), FFmpeg will by default extract only the first audio stream. Use -map 0:a:1 to select the second track if needed.