Extract Audio from M2TS to AMR — Free Online Tool

Extract audio from M2TS Blu-ray and AVCHD files and convert it to AMR format using the libopencore_amrnb codec — optimized for speech and voice telephony at very low bitrates. Ideal for pulling spoken dialogue or commentary tracks from high-definition video and compressing them into a mobile-friendly, bandwidth-efficient format.

FFmpeg Command

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Estimated output:

Conversion Complete!

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How It Works

M2TS files are MPEG-2 Transport Stream containers typically carrying multi-channel audio encoded as AAC, AC-3, DTS, or TrueHD alongside high-definition H.264 or H.265 video. This tool strips the video stream entirely using the -vn flag, then decodes the first audio track found in the M2TS container and re-encodes it using libopencore_amrnb — the Adaptive Multi-Rate Narrowband codec — at 12,200 bps. AMR-NB is a narrowband codec engineered for speech intelligibility over limited bandwidth, operating at a fixed sample rate of 8,000 Hz and mono channel output. This means any stereo or surround audio from the Blu-ray source is downmixed to mono and resampled from its original high-fidelity sample rate (typically 48,000 Hz) down to 8,000 Hz during encoding. The result is a very small file tuned for voice content rather than music or full-spectrum audio.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg multimedia processing tool. In this browser-based tool, this runs via FFmpeg.wasm compiled to WebAssembly, executing entirely within your browser without sending your M2TS file to any server.
-i input.m2ts Specifies the input file — an M2TS container, the standard format for Blu-ray disc video and AVCHD camcorder recordings. FFmpeg parses the MPEG-2 Transport Stream to identify all video, audio, and subtitle streams inside.
-vn Disables video output entirely. Since M2TS files contain high-definition video streams (typically H.264 or H.265), this flag tells FFmpeg to ignore them and extract only the audio, which is essential for producing an audio-only AMR output file.
-c:a libopencore_amrnb Sets the audio encoder to libopencore_amrnb, the open-source implementation of the Adaptive Multi-Rate Narrowband codec. This re-encodes the decoded M2TS audio (which may have been AAC, AC-3, or DTS) into the AMR-NB format optimized for speech at 8,000 Hz mono.
-b:a 12200 Sets the AMR-NB encoding bitrate to 12,200 bits per second, which corresponds to AMR mode 7 — the highest quality mode available for AMR-NB. This value must be one of the fixed AMR-NB mode bitrates; it is specified in bps (not kbps) because AMR operates at sub-kilobit granularity.
output.amr The output filename with the .amr extension. FFmpeg recognizes this extension and writes a standard 3GPP AMR file header, making the output compatible with mobile devices, telephony systems, and media players that support AMR-NB playback.

Common Use Cases

  • Extracting spoken commentary tracks from AVCHD camcorder recordings to send as voice notes over mobile networks with strict bandwidth limits
  • Pulling dialogue-only audio from Blu-ray rips for use in voice recognition or transcription pipelines that expect low-bitrate speech input
  • Converting director's commentary or audio description tracks from M2TS recordings into AMR files for playback on older mobile handsets or embedded telephony devices
  • Archiving interview or documentary narration audio from AVCHD footage in a compact, speech-optimized format for long-term storage at minimal file size
  • Preparing voice-over segments recorded on AVCHD cameras for upload to VoIP or IVR systems that require AMR-encoded audio input
  • Extracting a single language audio track from a multi-audio-track M2TS broadcast recording for distribution over low-bandwidth mobile connections

Frequently Asked Questions

Yes — this is a significant lossy conversion with noticeable quality reduction, by design. M2TS files typically carry audio at 48,000 Hz sample rate with full stereo or surround channels, while AMR-NB operates at a fixed 8,000 Hz sample rate in mono. This means most of the audio frequency range (above 4,000 Hz) is discarded. AMR is engineered specifically for speech intelligibility, so voice and dialogue remain understandable, but music, ambient sound, and cinematic audio will sound noticeably thin and degraded. Use this conversion only when the source audio is primarily speech.
By default, FFmpeg selects the first audio stream it finds in the M2TS container, which is typically the primary language or the default-flagged track. M2TS files from Blu-ray discs often contain multiple tracks (e.g., English DTS-HD, French AC-3, Spanish AAC). If you want to extract a specific track, you can add a stream selector like -map 0:a:1 to the command to target the second audio stream (zero-indexed). Without -map, you get whatever FFmpeg identifies as the best default audio stream.
Yes, AMR Wideband (libopencore_amrwb) operates at 16,000 Hz sample rate instead of 8,000 Hz, which significantly improves speech clarity and naturalness. To use it, change the codec flag in the command to -c:a libopencore_amrwb and use a compatible bitrate such as 23850. The output file will still use the .amr extension, though some systems distinguish AMR-WB files with a .awb extension. If your target device or system supports AMR-WB, it is the better choice for voice quality.
Replace the value after -b:a in the command with one of the valid AMR-NB bitrate modes: 4750, 5150, 5900, 6700, 7400, 7950, 10200, or 12200 (bits per second, not kilobits). The default 12200 bps is the highest quality AMR-NB mode and the most widely compatible. Lower values like 4750 bps produce even smaller files but with noticeably reduced intelligibility. Unlike most audio codecs, AMR-NB does not accept arbitrary bitrates — you must use one of these exact fixed mode values.
The size reduction is dramatic. A typical M2TS file encodes audio at 192–1,500 kbps (depending on whether it's AAC, AC-3, or lossless TrueHD), while AMR-NB at 12,200 bps is encoding at roughly 12 kbps. An hour of M2TS audio that might occupy 500 MB alongside the video, or 60–80 MB as a standalone audio track, will become roughly 5–6 MB in AMR-NB. The video stream is also stripped entirely, so the size difference from the original M2TS is enormous.
Yes, you can adapt the command for batch processing in a shell script. On Linux or macOS, use: for f in *.m2ts; do ffmpeg -i "$f" -vn -c:a libopencore_amrnb -b:a 12200 "${f%.m2ts}.amr"; done. On Windows Command Prompt: for %f in (*.m2ts) do ffmpeg -i "%f" -vn -c:a libopencore_amrnb -b:a 12200 "%~nf.amr". The in-browser tool processes one file at a time, but the FFmpeg command displayed on this page is particularly useful for batch workflows involving large collections of AVCHD or Blu-ray rips.

Technical Notes

AMR-NB produced by libopencore_amrnb is strictly mono and 8,000 Hz — FFmpeg will automatically handle the downmix from any M2TS audio layout (stereo, 5.1 surround, 7.1) and resample from 48,000 Hz or 44,100 Hz to 8,000 Hz before encoding. Metadata from the M2TS container such as language tags, title, or track names is not carried over to the AMR output, as the AMR format has no meaningful container-level metadata support. If your M2TS source uses a codec that requires a proprietary license (such as TrueHD or DTS-HD MA), FFmpeg will decode it during processing but you should ensure your FFmpeg build includes the necessary decoders — the WebAssembly build used in this browser tool includes standard decoders for AAC, AC-3, and DTS. The libopencore_amrnb encoder implements the 3GPP AMR-NB standard (GSM 06.90 / ETSI TS 126.090), and the resulting .amr files are compatible with 3GPP-compliant mobile devices, many VoIP platforms, and Android and iOS native audio players. Chapter and subtitle data from the M2TS source is discarded entirely since AMR does not support these features.

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