Convert MXF to AMR — Free Online Tool
Convert MXF broadcast footage to AMR audio, extracting and transcoding the audio stream using the libopencore_amrnb codec — ideal for pulling speech or voice content from professional production files into a mobile-friendly, low-bitrate format. AMR's speech-optimized compression makes it a practical target when you need lightweight voice audio from metadata-rich broadcast sources.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your MXF file here
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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
MXF files typically contain one or more audio tracks encoded in PCM (uncompressed, such as pcm_s16le or pcm_s24le) alongside video streams. During this conversion, FFmpeg discards the video entirely and extracts the audio, then transcodes it through the libopencore_amrnb encoder into Adaptive Multi-Rate Narrowband format. AMR-NB operates at a fixed 8kHz sample rate with 8-bit mono audio, so any high-fidelity PCM or AAC audio in the MXF source is downsampled and mixed to mono during encoding. The output is a highly compressed, speech-optimized .amr file at 12,200 bps — the highest standard AMR-NB bitrate, preserving as much speech intelligibility as the codec allows.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles all media decoding, processing, and encoding. In the browser, this runs as FFmpeg.wasm via WebAssembly — no installation required. |
-i input.mxf
|
Specifies the input MXF file. FFmpeg will parse the MXF container, identifying all available streams — including video (H.264, MPEG-2, or MJPEG), audio (PCM, AAC), and embedded broadcast metadata such as timecodes. |
-c:a libopencore_amrnb
|
Sets the audio encoder to libopencore_amrnb, which implements the AMR Narrowband codec. This transcodes the MXF audio (typically 48kHz PCM) into AMR-NB's speech-optimized format at 8kHz mono — the only configuration AMR-NB supports. |
-b:a 12200
|
Sets the AMR-NB audio bitrate to 12,200 bits per second, which corresponds to the highest quality AMR-NB encoding mode (Mode 7). This maximizes speech intelligibility within the constraints of the AMR-NB codec. Standard AMR-NB bitrates range from 4,750 to 12,200 bps. |
output.amr
|
Defines the output filename and triggers FFmpeg to write an AMR container. Because no video stream flags are set, FFmpeg automatically omits the video from the MXF source and writes only the transcoded AMR-NB audio stream to this file. |
Common Use Cases
- Extracting a reporter's on-location voiceover track from a broadcast MXF file for playback on a mobile telephony or voice messaging system
- Converting interview audio recorded on a professional broadcast camera to AMR for integration with a legacy telecom or IVR platform that only accepts AMR-NB
- Pulling dialogue or narration tracks from an MXF editing master to create compact AMR files for review on mobile devices with limited storage
- Repurposing speech content from broadcast production MXF archives into AMR format for use in mobile app audio assets or voice prompt libraries
- Reducing file size dramatically for rough voice approval — sending a low-bitrate AMR clip from a large MXF master for quick client sign-off via messaging apps that support AMR playback
Frequently Asked Questions
No. AMR is a single-channel (mono), single-track format, so only one audio stream will be carried to the output. By default, FFmpeg will select the first audio stream from the MXF file and mix it down to mono at 8kHz. If your MXF contains multiple tracks — such as separate left/right dialogue channels or a multi-track stem mix — you would need to specify which track to extract using the '-map' flag before running the conversion.
AMR Narrowband (AMR-NB) encodes audio at a fixed 8kHz sample rate, which means frequencies above 4kHz are discarded entirely during downsampling. Professional MXF files often contain 48kHz PCM audio with a full frequency range — broadcast-quality sound. The conversion is intentionally aggressive in compression because AMR was designed for voice telephony, not music or full-bandwidth audio. The format excels at preserving speech intelligibility at very low bitrates but is unsuitable for any audio requiring high fidelity.
Replace the '-b:a 12200' value with any of the standard AMR-NB bitrates: 4750, 5150, 5900, 6700, 7400, 7950, or 10200 (all in bits per second). For example, '-b:a 7950' gives a mid-quality setting that is still quite intelligible for speech. The default in this tool is 12200, which is the highest AMR-NB mode and produces the best speech quality the codec can deliver. Lower values will reduce file size further but may introduce more compression artifacts in voice recordings.
No. AMR is a minimal audio container with no support for timecodes, production metadata, or the rich descriptive fields that MXF carries for broadcast workflows. All SMPTE timecode data, reel names, and embedded descriptive metadata in the MXF file will be discarded during this conversion. If preserving that metadata matters, you should archive the original MXF file — AMR is suitable only as a distribution or review format for the extracted speech audio.
Yes, with a small modification in your terminal. On Linux or macOS, you can use a shell loop: 'for f in *.mxf; do ffmpeg -i "$f" -c:a libopencore_amrnb -b:a 12200 "${f%.mxf}.amr"; done'. On Windows Command Prompt, use: 'for %f in (*.mxf) do ffmpeg -i "%f" -c:a libopencore_amrnb -b:a 12200 "%~nf.amr"'. This processes each MXF file in the current directory and outputs a matching .amr file. The browser-based tool on this page handles one file at a time, so the FFmpeg command is especially useful for batch workflows on large collections.
If your target platform supports it, AMR Wideband (AMR-WB, codec: libopencore_amrwb) offers significantly better audio quality by encoding at 16kHz instead of 8kHz — capturing more of the speech frequency range. To use it, change the command to '-c:a libopencore_amrwb' and use a compatible wideband bitrate like 23850. However, AMR-WB is less universally supported than AMR-NB, particularly on older telecom systems and some messaging apps. For maximum compatibility with legacy mobile or telephony systems, AMR-NB at 12200 bps remains the safer default.
Technical Notes
MXF (Material Exchange Format) is a professional broadcast container commonly storing uncompressed or lightly compressed audio — typically 16-bit or 24-bit PCM at 48kHz — alongside video encoded in H.264, MPEG-2, or MJPEG. AMR-NB (Adaptive Multi-Rate Narrowband), implemented here via the libopencore_amrnb library, operates under strict constraints: mono audio only, 8kHz sample rate, and bitrates capped at 12,200 bps. This means FFmpeg must perform significant downsampling (from 48kHz to 8kHz) and channel downmixing (from stereo or multi-channel to mono) as part of the transcode — a lossy process with no workaround given AMR-NB's codec specification. The resulting file will be dramatically smaller than the source MXF (often by a factor of 50–100x), but this comes at the cost of all non-speech audio content. The AMR format does not support chapters, subtitles, or video, and the container carries no meaningful metadata fields. This conversion is most appropriate when the MXF source contains isolated dialogue or narration and the destination is a speech-aware platform such as a voice response system, mobile messaging app, or telephony integration. Users processing files larger than 1GB should use the displayed FFmpeg command locally, as the browser-based tool is limited to 1GB inputs.