MyMediaTools

Convert MTS to CAF — Free Online Tool

Convert MTS camcorder footage (AVCHD with H.264 video and AC-3/AAC audio) to CAF, Apple's Core Audio Format, extracting the audio stream and encoding it as uncompressed PCM for high-fidelity use in macOS and iOS audio workflows. Ideal for pulling clean audio from Sony or Panasonic camcorder recordings into Logic Pro, GarageBand, or other Apple audio tools.

FFmpeg Command

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Estimated output:

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How It Works

MTS files are AVCHD containers carrying H.264 video and typically AC-3 or AAC audio, recorded directly by Sony and Panasonic camcorders. Since CAF is a pure audio container with no video support, the video stream is discarded entirely during this conversion — only the audio is processed. The AC-3 or AAC audio from the MTS file is decoded and then re-encoded as 16-bit signed little-endian PCM (pcm_s16le) inside the CAF container. This means the output is uncompressed linear audio, similar to a WAV or AIFF file but wrapped in Apple's CAF format which supports files larger than 4GB and integrates natively with Core Audio on macOS and iOS. The trade-off is significantly larger file size compared to the compressed AC-3 original, but with no further generational quality loss from codec compression.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg command-line tool, which handles all demuxing, decoding, encoding, and muxing for this conversion. In the browser version, this runs via FFmpeg.wasm compiled to WebAssembly, executing the same logic as the desktop binary.
-i input.mts Specifies the input file — an MTS file in the AVCHD format as recorded by Sony or Panasonic camcorders. FFmpeg will demux this MPEG-2 Transport Stream container and identify the H.264 video and AC-3/AAC audio streams inside it.
-c:a pcm_s16le Sets the audio codec to 16-bit signed little-endian PCM, which is uncompressed linear audio. This decodes the compressed AC-3 or AAC audio from the MTS source and stores it as raw audio samples in the CAF container, eliminating any further lossy compression while keeping full compatibility with Core Audio on macOS and iOS.
-b:a 128k Specifies a target audio bitrate of 128kbps. For PCM codecs like pcm_s16le this parameter is effectively ignored since PCM bitrate is fixed by bit depth and sample rate rather than a compression target — it is included here for consistency with the tool's interface but has no impact on the output file.
output.caf Defines the output file with the .caf extension, which signals FFmpeg to use Apple's Core Audio Format container. This container will hold the uncompressed PCM audio stream extracted from the MTS source, ready for direct use in Logic Pro, GarageBand, Final Cut Pro, or any other Core Audio-compatible application.

Common Use Cases

  • Importing dialogue or ambient sound recorded on a Sony or Panasonic camcorder directly into Logic Pro or GarageBand as a high-quality uncompressed audio track
  • Extracting interview audio from AVCHD camcorder footage to use as source material in a podcast production on macOS
  • Pulling clean location sound from MTS recordings into Final Cut Pro's audio pipeline via CAF for professional audio post-production
  • Converting camcorder-recorded music performances from MTS to uncompressed CAF for archival or further mastering in an Apple ecosystem studio
  • Stripping and converting the AC-3 or AAC audio from an AVCHD clip into CAF for use in an iOS app's audio asset pipeline where Core Audio compatibility is required
  • Preparing field-recorded audio captured on a camcorder for round-tripping into SoundTrack Pro or other Core Audio-native editors without introducing additional codec compression

Frequently Asked Questions

There is a one-time decode step where the original AC-3 or AAC audio from the MTS file is decompressed, and that decoded audio is then written as uncompressed PCM — so any quality loss baked into the original camcorder recording stays, but no new compression artifacts are introduced. The pcm_s16le encoding in the output CAF is lossless in the sense that it stores raw 16-bit audio samples with no further codec compression. If the original camcorder audio was AC-3 at 192kbps or AAC at 128kbps, you are capturing exactly what was encoded, just in an uncompressed form.
The MTS file stores its audio in a compressed codec — either AC-3 or AAC — which can represent audio at roughly 128–256kbps. The output CAF uses pcm_s16le, which is uncompressed 16-bit audio and typically runs at around 1,411kbps for stereo at 44.1kHz or higher for 48kHz camcorder audio. This means the audio portion alone can be 5–10x larger than the compressed original. The video stream from the MTS file is also discarded, so the CAF will only contain audio, but that audio will be substantially larger per second of content.
MTS/AVCHD files carry limited metadata, and the conversion process focuses on the audio stream rather than camcorder-specific metadata fields. CAF supports rich metadata including channel layout descriptors and markers, but FFmpeg's MTS-to-CAF conversion does not automatically map AVCHD recording metadata into CAF metadata chunks. You would need to add metadata manually using an audio editor or additional FFmpeg flags like -metadata if preserving specific fields is important for your workflow.
Yes — CAF is a native Apple format and is directly supported by Logic Pro, GarageBand, Final Cut Pro, and any application built on Core Audio on macOS or iOS. The pcm_s16le encoding used in the output is standard uncompressed audio that these applications will read without any conversion step. Simply drag the CAF file into your Logic Pro project or GarageBand track and it will import immediately as a clean audio region.
By default, FFmpeg selects the first audio stream from the MTS file, which is typically the main stereo or surround mix recorded by the camcorder. CAF does not support multiple audio tracks in a single file, so only one stream can be included. If your AVCHD footage has a secondary audio track (such as a dual-system scratch track), you can specify it explicitly in the FFmpeg command by adding -map 0:a:1 before the output filename to select the second audio stream instead of the default first.
The codec flag -c:a pcm_s16le specifies 16-bit little-endian PCM. To get 24-bit audio — useful if your camcorder recorded at 24-bit — replace it with -c:a pcm_s24le. To force a specific sample rate such as 48kHz (standard for camcorder audio) or 44.1kHz (CD standard), add -ar 48000 or -ar 44100 before the output filename. For example: ffmpeg -i input.mts -c:a pcm_s24le -ar 48000 output.caf gives you 24-bit, 48kHz uncompressed audio in the CAF container, which preserves the full resolution of professional AVCHD recordings.

Technical Notes

MTS files from Sony and Panasonic AVCHD camcorders typically carry audio as AC-3 (Dolby Digital) at 192–256kbps or AAC, sampled at 48kHz — the broadcast standard. When converting to CAF with pcm_s16le, FFmpeg decodes this compressed audio and writes 16-bit uncompressed samples, resulting in CD-quality or broadcast-quality audio depending on the source sample rate. CAF was designed by Apple specifically to overcome the 4GB file size limit of WAV and AIFF, making it well-suited for long-form camcorder recordings. The format natively integrates with macOS's Core Audio stack, meaning no third-party codec installation is needed on any modern Mac. One important limitation: CAF does not support multiple audio tracks in a single file, so if your AVCHD source carries both a main mix and a separate microphone channel, you will need to run separate conversions for each stream. The -b:a flag in the base command has no effect on PCM codecs since PCM audio is inherently uncompressed and bitrate is determined entirely by bit depth and sample rate — you can safely omit it or override the codec to a compressed option like aac or flac if file size is a concern.

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