Convert M4V to AMR — Free Online Tool
Convert M4V video files to AMR audio, extracting the speech or voice track and re-encoding it using the Adaptive Multi-Rate Narrowband codec (libopencore_amrnb) — the same codec used in mobile telephony. This is especially useful when you need to isolate spoken dialogue or voice content from iTunes videos or iOS-recorded clips in a compact, GSM-compatible format.
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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
M4V is a video container that typically holds H.264 or H.265 video alongside AAC audio. AMR, by contrast, is an audio-only format with no video support, so during this conversion the video stream is completely discarded. The AAC audio track is decoded and then re-encoded from scratch using the libopencore_amrnb encoder into AMR Narrowband format. AMR Narrowband operates at a fixed sample rate of 8 kHz and is engineered specifically for human speech — it is not a general-purpose audio codec and will compress music or complex audio with noticeable quality loss. The output file will be dramatically smaller than the original M4V, containing only a low-bitrate speech-optimized audio stream.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing engine. This is the same underlying engine running in your browser via WebAssembly for the in-browser conversion. |
-i input.m4v
|
Specifies the input file — an M4V container, which typically holds H.264 or H.265 video alongside an AAC audio track. FFmpeg will read both streams but only the audio will be used in the output. |
-c:a libopencore_amrnb
|
Sets the audio encoder to libopencore_amrnb, the open-source implementation of the AMR Narrowband codec. This re-encodes the AAC audio from the M4V from scratch into AMR NB format, resampling to 8 kHz in the process and discarding the video stream implicitly because AMR cannot contain video. |
-b:a 12200
|
Sets the AMR encoding mode to 12.2 kbps, which is the highest-quality AMR Narrowband mode (Mode 7, also called MR122). This is the standard rate used in GSM full-rate voice calls and provides the best speech intelligibility available within the AMR NB codec. |
output.amr
|
Defines the output filename with the .amr extension, which signals FFmpeg to write an IETF-formatted AMR Narrowband file — the standard single-channel AMR container recognized by mobile phones and telephony applications. |
Common Use Cases
- Extracting spoken dialogue or narration from an iTunes lecture or educational video to load onto a GSM-compatible voice recorder or older mobile handset
- Preparing voice-over audio from an iOS-recorded M4V clip for submission to a telephony or IVR system that requires AMR-encoded audio files
- Stripping the audio track from an M4V podcast video to create a compact AMR file for playback on feature phones with limited storage
- Converting spoken-word iTunes audiobook video content into AMR format for compatibility with older Nokia or Symbian-based devices
- Archiving interview or voice-memo video recordings from Apple devices into AMR format, which is the native storage format used by many mobile phone call recorders
- Extracting a single-speaker commentary track from an M4V video to test AMR codec behavior for a mobile VoIP or telephony development project
Frequently Asked Questions
No — there will be a significant quality reduction, which is inherent to AMR Narrowband itself rather than a flaw in the conversion. AMR NB resamples audio to 8 kHz, which means all frequencies above 4 kHz are discarded. The original M4V likely contains AAC audio sampled at 44.1 kHz or 48 kHz, so music, sound effects, and any high-frequency content will sound dull and telephone-like in the output. AMR is engineered exclusively for the frequency range of human speech, so spoken dialogue will remain intelligible, but it is not suitable for music or general audio.
The video stream is entirely dropped. AMR is a pure audio-only format with no container support for video tracks, so FFmpeg cannot carry the video into the output file. Only the first audio track from the M4V is decoded and re-encoded as AMR. If your M4V contains multiple audio tracks (for example, different language dubs), only the default track is used unless you explicitly specify a different stream with the -map flag.
AMR uses a set of fixed codec modes defined by the GSM and 3GPP telephony standards, and these modes are specified in raw bits per second rather than the kilobits-per-second convention common in MP3 or AAC encoding. The default mode used here, 12200 bps (12.2 kbps), is the highest-quality AMR Narrowband mode and the one most commonly used in voice calls. The other modes — ranging from 4750 bps up to 12200 bps — all operate at 8 kHz sample rate; the bitrate difference affects how aggressively the speech is compressed, not the sample rate.
Yes. AMR Wideband (AMR-WB, using the libopencore_amrwb encoder) operates at 16 kHz instead of 8 kHz, capturing a wider speech frequency range and sounding noticeably clearer. To use it, change the command to: ffmpeg -i input.m4v -c:a libopencore_amrwb -b:a 23850 output.awb — note that AMR-WB files conventionally use the .awb extension. However, AMR-WB compatibility is narrower than AMR-NB; many older devices and telephony systems only support Narrowband.
Replace the 12200 value in -b:a 12200 with any of the valid AMR NB mode bitrates: 4750, 5150, 5900, 6700, 7400, 7950, or 10200. For example, ffmpeg -i input.m4v -c:a libopencore_amrnb -b:a 7950 output.amr will encode at 7.95 kbps, which still produces intelligible speech at a smaller file size. FFmpeg will snap to the nearest valid AMR mode if you enter a value that does not exactly match one of these fixed rates.
Yes. On Linux or macOS you can run: for f in *.m4v; do ffmpeg -i "$f" -c:a libopencore_amrnb -b:a 12200 "${f%.m4v}.amr"; done — this loops through every M4V in the current directory and produces a matching AMR file. On Windows Command Prompt, use: for %f in (*.m4v) do ffmpeg -i "%f" -c:a libopencore_amrnb -b:a 12200 "%~nf.amr". Note that AMR does not support chapters or metadata tags in the way M4V does, so any iTunes metadata in the source files will not carry over.
Technical Notes
The libopencore_amrnb encoder used in this conversion is an open-source implementation of the 3GPP AMR Narrowband speech codec (standardized in GSM 06.90). Because AMR NB is a speech codec, it applies voice activity detection and comfort noise generation — mechanisms that are beneficial for phone calls but can cause artifacts when encoding music or mixed audio from an M4V source. The output .amr file uses the single-channel IETF AMR file format with a '#!AMR\n' magic header, which is compatible with most mobile devices and telephony systems but is distinct from the 3GP container sometimes used to wrap AMR streams. Metadata carried in the M4V container — including iTunes tags, chapter markers, and multiple audio track labels — cannot be preserved in AMR, as the format supports none of these features. If your M4V source was DRM-protected (FairPlay), FFmpeg will be unable to decode the audio stream and the conversion will fail; the file must be DRM-free. The output file size will be extremely small relative to the M4V — a one-hour M4V that might be 1.5 GB will produce an AMR file of roughly 5–6 MB at 12.2 kbps.