Extract Audio from MKV to AMR — Free Online Tool
Extract audio from MKV video files and convert it to AMR format using the libopencore_amrnb codec — a speech-optimized, low-bitrate format originally designed for mobile telephony. This tool is ideal for pulling voice recordings or dialogue from MKV files into a compact AMR file compatible with older mobile devices and telephony systems.
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FFmpeg Command
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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
During this conversion, FFmpeg discards the MKV container's video stream entirely and re-encodes the audio stream using the libopencore_amrnb codec, producing an AMR-NB (Narrowband) file. Unlike a simple remux, this is a full audio transcode — the original audio (which in an MKV is commonly AAC, Opus, or FLAC) is decoded and then re-encoded at AMR's fixed 8kHz sample rate and a bitrate of 12,200 bps. AMR-NB uses a lossy, perceptual compression model tuned specifically for human speech frequencies, so music or broadband audio will lose significant fidelity, but spoken voice can remain intelligible at very small file sizes.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool, which handles all demuxing, decoding, encoding, and muxing steps required to convert the MKV audio stream into an AMR-NB file. |
-i input.mkv
|
Specifies the input Matroska file, which may contain video, multiple audio tracks (commonly AAC or Opus by default), subtitles, and chapter data — all of which FFmpeg will parse from the MKV container. |
-vn
|
Disables video output entirely, ensuring that the MKV's video stream is dropped and only the audio stream is processed — essential since AMR is a pure audio format with no ability to carry video data. |
-c:a libopencore_amrnb
|
Selects the libopencore_amrnb encoder to re-encode the audio into AMR Narrowband format, operating at a fixed 8kHz sample rate — the codec used in mobile telephony and the standard encoder for producing .amr files. |
-b:a 12200
|
Sets the AMR-NB encoding bitrate to 12,200 bits per second, which corresponds to AMR-NB Mode 7 — the highest quality mode available in the AMR-NB specification, giving the best speech intelligibility within the codec's constraints. |
output.amr
|
Defines the output filename with the .amr extension, which tells FFmpeg to write a raw AMR bitstream file using the RFC 4867 storage format header recognized by mobile devices and telephony systems. |
Common Use Cases
- Converting MKV recordings of phone interviews or voice memos into AMR for storage on legacy mobile devices that only support AMR playback
- Extracting spoken dialogue from an MKV lecture or conference recording to create a compact AMR file for telephony or IVR system integration
- Preparing voice audio from MKV screen recordings for upload to systems that require AMR-formatted voice clips, such as certain VoIP or PBX platforms
- Archiving MKV-captured field interviews in AMR format to reduce storage footprint when audio quality requirements are minimal and speech intelligibility is sufficient
- Stripping the audio track from a dashcam or bodycam MKV footage into AMR for use in transcription pipelines that accept AMR input
- Batch-converting MKV podcast source files into AMR for distribution on platforms or devices with strict bandwidth or storage limitations
Frequently Asked Questions
Yes — AMR-NB encodes at a fixed 8kHz sample rate and is engineered exclusively for speech, so any music, sound effects, or broadband audio in your MKV will sound noticeably degraded or muffled. Spoken voice, however, can remain reasonably intelligible even at AMR's lowest bitrates. If your MKV contains music or high-fidelity audio and you need good quality output, AMR is not the right target format.
All of them are lost. AMR is a bare audio format with no container features — it cannot store subtitles, chapter markers, metadata, or secondary audio tracks. FFmpeg selects the default (first) audio stream from the MKV and encodes only that stream into the AMR file. If your MKV has multiple audio tracks and you need a specific one, you would add a flag like '-map 0:a:1' to select the second audio track before running the command.
AMR-NB (Narrowband) operates at 8kHz and bitrates from 4,750 to 12,200 bps, while AMR-WB (Wideband) operates at 16kHz with higher bitrates, delivering noticeably better speech quality. This tool uses libopencore_amrnb and produces AMR-NB files, which have the broadest compatibility with older mobile phones and telephony hardware. If you need AMR-WB for better speech fidelity, you would need to modify the FFmpeg command to use libopencore_amrwb instead.
Replace '12200' in the '-b:a 12200' flag with one of AMR-NB's supported fixed bitrates: 4750, 5150, 5900, 6700, 7400, 7950, 10200, or 12200 (all in bits per second). Unlike most audio codecs, AMR-NB only accepts these exact mode values — arbitrary bitrates are not supported. Using 12200 bps gives the highest quality and largest file size, while 4750 bps gives the smallest file at the cost of voice clarity.
Yes. On Linux or macOS you can use a shell loop: 'for f in *.mkv; do ffmpeg -i "$f" -vn -c:a libopencore_amrnb -b:a 12200 "${f%.mkv}.amr"; done'. On Windows Command Prompt, use 'for %f in (*.mkv) do ffmpeg -i "%f" -vn -c:a libopencore_amrnb -b:a 12200 "%~nf.amr"'. This is especially useful for files over 1GB, which exceed the browser tool's limit but run efficiently on the desktop with the same command.
AMR is not a container — it is a raw bitstream format that only holds AMR-encoded audio. It cannot carry AAC, MP3, Opus, FLAC, or any other codec that an MKV might contain. This means FFmpeg must fully decode the original MKV audio stream and then re-encode it from scratch using libopencore_amrnb, which is why this conversion is lossy regardless of the source audio quality.
Technical Notes
AMR-NB (Narrowband) imposes strict technical constraints that are important to understand before converting. The codec enforces an 8kHz sample rate and mono channel output — if your MKV contains stereo or surround audio, FFmpeg will automatically downmix to mono and downsample to 8kHz as part of the transcode. The eight supported bitrate modes (4,750–12,200 bps) are fixed by the AMR-NB specification and are not interpolatable. The libopencore_amrnb encoder used by FFmpeg is based on the 3GPP reference implementation and is well-tested for telephony use cases. Note that AMR files carry virtually no metadata — there is no standard container layer for embedding title, artist, or album tags, so all MKV metadata is discarded. AMR-NB files use the .amr extension and are recognized by the RFC 4867 storage format header. Compatibility is strongest with Android devices, older Nokia and feature phones, and VoIP/IVR platforms; desktop media players may require a specific codec or plugin to play .amr files. If AMR-WB (libopencore_amrwb, 16kHz, .awb extension) is acceptable for your use case, it offers meaningfully better speech quality.