Convert MTS to AIFC — Free Online Tool
Convert MTS camcorder footage (AVCHD with H.264 video and AC-3/AAC audio) to AIFC, extracting and repackaging the audio track as uncompressed PCM in Apple's compressed AIFF container. This is ideal for pulling broadcast-quality audio from Sony or Panasonic camcorder recordings into a professional audio workflow.
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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 video and audio in an MPEG-2 Transport Stream container, typically with H.264 video and either AC-3 or AAC audio. During this conversion, FFmpeg discards the H.264 video stream entirely and decodes the audio track, then re-encodes it as 16-bit big-endian PCM (pcm_s16be) wrapped in the AIFC container. AIFC is Apple's extension of AIFF that supports both uncompressed and compressed audio — in this case the output uses uncompressed PCM, making it lossless at the PCM stage. However, if the original MTS audio was AC-3 or AAC (both lossy formats), the AIFC output reflects that prior lossy encoding; no audio data lost in the original compression is recovered. The result is a standalone AIFC file with no video, compatible with professional audio software on macOS and other platforms that support the AIFF/AIFC format.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles decoding the AVCHD transport stream and encoding the AIFC output. In this browser tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) with no server involvement. |
-i input.mts
|
Specifies the input MTS file — an AVCHD MPEG-2 Transport Stream from a Sony or Panasonic camcorder, typically containing H.264 video and AC-3 or AAC audio. FFmpeg will demux this container to access its individual audio and video streams. |
-c:a pcm_s16be
|
Sets the output audio codec to 16-bit signed big-endian PCM, the native uncompressed audio format for AIFC. This decodes the lossy AC-3 or AAC audio from the MTS file and re-encodes it as uncompressed PCM, producing an editable, DAW-ready audio file without further lossy compression. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps, but for PCM codecs like pcm_s16be this parameter has no effect — PCM bitrate is fixed by sample rate and bit depth (typically ~1536 kbps for 48 kHz stereo 16-bit from an AVCHD source). This flag is a tool default and can be omitted when running the command locally. |
output.aifc
|
Defines the output filename and tells FFmpeg to use the AIFC container format. FFmpeg infers the container from the .aifc extension and wraps the pcm_s16be audio stream in Apple's compressed AIFF container structure, producing a file compatible with Logic Pro, Pro Tools, and other professional audio applications. |
Common Use Cases
- Importing dialogue or ambient audio recorded on a Sony or Panasonic camcorder into Logic Pro or Final Cut Pro for professional audio editing without carrying the video stream.
- Archiving the audio track from AVCHD event recordings (weddings, conferences, concerts) as a PCM AIFC file for long-term storage in a format independent of video codecs.
- Preparing camcorder-recorded voiceovers or interviews for delivery to an audio post-production house that requires AIFF/AIFC format source files.
- Extracting a clean audio bed from MTS footage to use as a music or sound effects reference track in a Digital Audio Workstation (DAW) during video scoring.
- Converting field recordings captured on an AVCHD camcorder to AIFC so they can be ingested into Apple's audio toolchain (GarageBand, Logic, Audio MIDI Setup) without format conversion errors.
- Pulling the audio channel from multi-camera MTS files to sync and align tracks in an audio editor before the final video edit.
Frequently Asked Questions
No — the conversion cannot restore audio quality that was lost during the original AVCHD recording. MTS files typically carry AC-3 or AAC audio, both of which are lossy formats. FFmpeg decodes that lossy audio and then writes it as uncompressed PCM in the AIFC file, which means the AIFC is a lossless representation of the already-compressed audio. You get a clean, editable PCM file, but the artifacts introduced during the original camcorder encoding remain.
The MTS file stores audio as compressed AC-3 or AAC, which can be 10–20 times smaller than uncompressed audio. The AIFC output uses pcm_s16be — uncompressed 16-bit PCM — so the file size grows dramatically because every audio sample is stored at full resolution without compression. For example, a 1-hour MTS with 192 kbps AC-3 audio might yield an AIFC file of roughly 600 MB for the audio alone. This is expected and is the nature of converting from compressed to uncompressed PCM.
AIFC has limited metadata support compared to modern containers. FFmpeg will attempt to map common metadata tags (like title or date) from the MTS transport stream into the AIFC file, but camera-specific AVCHD metadata — such as GPS coordinates, scene mode, or camera model stored in the MTS stream — is generally not preserved. If metadata retention is critical, consider keeping the original MTS file as the archive master and using the AIFC only as the working audio file.
Yes. The default command uses pcm_s16be (16-bit signed big-endian PCM), but AIFC also supports pcm_s24be (24-bit) and pcm_s32be (32-bit), as well as floating-point formats like pcm_f32be. To get 24-bit output, change the codec flag in the FFmpeg command to '-c:a pcm_s24be'. This won't recover detail lost by the original lossy MTS audio, but 24-bit is a common delivery requirement in professional audio post-production workflows.
You can loop over your MTS files in a shell script. On Linux or macOS, run: 'for f in *.mts; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.mts}.aifc"; done'. On Windows Command Prompt, use: 'for %f in (*.mts) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aifc"'. Each MTS file will be processed sequentially and produce a matching AIFC file in the same directory.
AIFC originated as an Apple format, but it is widely supported across professional audio applications on multiple platforms, including Adobe Audition, Pro Tools, Reaper, and most DAWs. However, consumer-focused software and web players may not support AIFC as reliably as WAV or MP3. If broad compatibility outside professional audio tools is a concern, consider using AIFF or WAV as the target format instead, which share the same PCM audio quality but have wider support.
Technical Notes
The AIFC container uses big-endian byte ordering (pcm_s16be), which is native to the format's Apple and Motorola 68k heritage — this is distinct from WAV, which stores PCM in little-endian format (pcm_s16le). Because MTS is a transport stream format designed for broadcast and camcorder use, it may contain multiple audio tracks (e.g., stereo plus a scratch track), but AIFC does not support multiple audio tracks. FFmpeg will by default select the first (or best-ranked) audio stream from the MTS file; if you need a specific audio track, add '-map 0:a:1' (or the relevant index) to the command. Subtitle streams embedded in the MTS file are silently dropped, as AIFC has no subtitle support. The '-b:a 128k' flag in the command has no practical effect on PCM codecs like pcm_s16be, since PCM bitrate is determined entirely by sample rate and bit depth — it can be safely omitted when running the command locally. For a 48 kHz stereo 16-bit PCM stream (typical for AVCHD footage), the actual bitrate will be approximately 1536 kbps regardless of the '-b:a' setting.