Extract Audio from MTS to AIFC — Free Online Tool

Extract audio from MTS camcorder footage and save it as an AIFC file with uncompressed 16-bit big-endian PCM audio. This tool strips the H.264 video and AC-3/AAC audio streams from the AVCHD container and re-encodes the audio into Apple's professional AIFC format — ideal for post-production workflows on macOS and in DAWs like Logic Pro.

FFmpeg Command

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Estimated output:

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How It Works

MTS files from Sony and Panasonic camcorders store audio as either AC-3 (Dolby Digital) or AAC inside an MPEG-2 Transport Stream container. During this conversion, FFmpeg discards the H.264 video stream entirely and decodes the compressed audio track, then re-encodes it as PCM_S16BE — signed 16-bit big-endian pulse-code modulation — wrapped in an AIFC container. Because the source audio is lossy (AC-3 or AAC) and the output is uncompressed PCM, the resulting AIFC file preserves the full decoded audio signal with no further lossy compression, though it cannot recover quality that was already lost during the original camcorder recording. The AIFC big-endian format is native to Apple's audio ecosystem and can be opened directly by Logic Pro, GarageBand, and most professional macOS audio tools.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg binary. In the browser-based tool, this runs via FFmpeg.wasm (WebAssembly) entirely on your local machine — no data is sent to a server. On your desktop, this calls your locally installed FFmpeg executable.
-i input.mts Specifies the input MTS file — an AVCHD MPEG-2 Transport Stream from a Sony or Panasonic camcorder, containing an H.264 video stream and compressed AC-3 or AAC audio.
-vn Disables video output entirely, telling FFmpeg to ignore the H.264 video stream in the MTS file. This is essential for audio-only extraction — without this flag, FFmpeg would attempt to include video in the output, which AIFC does not support.
-c:a pcm_s16be Sets the audio codec to signed 16-bit big-endian PCM, the native uncompressed format for AIFC files. The camcorder's compressed AAC or AC-3 audio is fully decoded and re-encoded as raw PCM, producing an uncompressed audio file suitable for editing in Logic Pro, GarageBand, or any macOS DAW.
-b:a 128k Nominally sets a target audio bitrate of 128 kbps, but this flag has no effect when the codec is an uncompressed PCM variant like pcm_s16be — the actual bitrate is determined entirely by the sample rate and bit depth. It is included by default for consistency with the tool's interface but can be safely omitted for this specific conversion.
output.aifc Specifies the output filename with the .aifc extension, which tells FFmpeg to use the AIFC container format. The .aifc extension distinguishes compressed or big-endian PCM AIFF variants from standard .aiff files and ensures correct handling by macOS audio applications.

Common Use Cases

  • Importing dialogue or ambient audio from Sony or Panasonic camcorder footage into Logic Pro or Pro Tools for professional audio post-production without carrying the video stream
  • Extracting a clean audio track from AVCHD interview footage to hand off to an audio editor who does not have video editing software
  • Archiving the audio portion of camcorder recordings as uncompressed PCM in AIFC format for long-term preservation in a macOS-based media library
  • Pulling the audio from event or wedding footage shot on an AVCHD camcorder to sync manually with a separate multi-track audio recording in a DAW
  • Separating the audio track from MTS clips before editing in Final Cut Pro so the audio can be processed, noise-reduced, or sweetened independently
  • Converting MTS audio to AIFC as an intermediate step before further processing or re-encoding to a broadcast-ready format like WAV or AIFF in a macOS post-production pipeline

Frequently Asked Questions

Not exactly — the AIFC file will be an uncompressed PCM representation of whatever the AC-3 or AAC decoder produces, so no additional lossy compression is applied. However, the quality ceiling is set by the original camcorder recording: if the MTS file used AAC at 128k or AC-3 at 192k, those artifacts are already baked into the audio signal. The AIFC output is bit-for-bit stable from that decoded signal forward, making it suitable for editing and further processing without any generational quality loss.
The default codec chosen is PCM_S16BE (signed 16-bit big-endian PCM), which matches the standard CD-quality bit depth and is the most universally compatible option for AIFC across macOS applications and DAWs. Since the source audio in MTS files from consumer and prosumer camcorders is typically recorded at 16-bit depth (either in AC-3 or AAC), upsizing to 24-bit or 32-bit would not add real information. If your camcorder recorded at higher bit depths or your DAW requires 24-bit, you can change the codec in the FFmpeg command to pcm_s24be.
To increase bit depth, replace pcm_s16be with pcm_s24be or pcm_s32be in the command: ffmpeg -i input.mts -vn -c:a pcm_s24be output.aifc. For PCM codecs, the -b:a bitrate flag has no practical effect because the bitrate is determined entirely by the sample rate and bit depth of uncompressed audio — you control quality by choosing the PCM variant, not a bitrate. If you want floating-point precision for mixing, use pcm_f32be instead.
By default, FFmpeg selects the best audio stream (usually the first or highest-channel-count track) and writes only that one to the AIFC output, since AIFC does not support multiple audio tracks. If you need a specific track, add -map 0:a:1 (for the second audio stream) to the command before the output filename: ffmpeg -i input.mts -vn -map 0:a:1 -c:a pcm_s16be output.aifc. You can inspect all streams in the MTS file by running ffmpeg -i input.mts before conversion.
MTS files embed metadata such as recording timestamps and device information in the MPEG-2 Transport Stream headers, but the AIFC format has limited metadata support compared to the source container. FFmpeg will attempt to copy common tags (like creation time) where the AIFC format allows, but camcorder-specific metadata such as GPS coordinates, timecode, or scene data from the AVCHD structure will not be transferred. If metadata preservation is critical, consider keeping the original MTS file alongside the extracted AIFC.
Yes — on macOS or Linux you can use a shell loop: for f in *.mts; do ffmpeg -i "$f" -vn -c:a pcm_s16be -b:a 128k "${f%.mts}.aifc"; done. On Windows Command Prompt, use: for %f in (*.mts) do ffmpeg -i "%f" -vn -c:a pcm_s16be -b:a 128k "%~nf.aifc". This is especially useful for processing large batches of AVCHD footage from a camcorder memory card, and for files over 1GB where the browser-based tool's limit applies.

Technical Notes

MTS (AVCHD) audio is typically encoded as AAC (in newer camcorder models) or AC-3/Dolby Digital (in older Sony and Panasonic models), usually at 48 kHz sample rate — note that the FFmpeg command preserves the original sample rate of the source, so the AIFC output will be at 48 kHz rather than 44.1 kHz unless you explicitly add -ar 44100. The AIFC container is Apple's extension of AIFF and stores big-endian PCM data, making it natively readable on macOS without any codec installation. Unlike WAV (which uses little-endian byte order), AIFC's big-endian PCM (PCM_S16BE) means the file is not directly interchangeable with WAV on all platforms — Windows applications may not handle it without conversion. The -b:a 128k flag in the command is technically ignored for lossless PCM codecs like pcm_s16be, since uncompressed audio bitrate is a fixed function of sample rate × bit depth × channels (e.g., 48000 Hz × 16 bits × 2 channels = 1536 kbps). File sizes will be substantially larger than the source MTS audio: a one-hour stereo recording at 48 kHz 16-bit PCM produces approximately 660 MB of audio data in the AIFC file, compared to perhaps 50–100 MB for the same duration in AC-3 or AAC inside the original MTS.

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