Extract Audio from AVI to AMR — Free Online Tool
Extract audio from AVI video files and convert it to AMR format using the libopencore_amrnb codec — a speech-optimized, low-bitrate format originally designed for mobile telephony. Ideal for repurposing recorded speech or dialogue from legacy AVI files into a format compatible with mobile voice systems.
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FFmpeg Command
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Estimated output:
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DownloadHow It Works
During this conversion, FFmpeg discards the video stream entirely and re-encodes the audio from the AVI container using the libopencore_amrnb codec, writing the result to an AMR file. AVI files commonly carry audio in MP3 (libmp3lame) format, which must be fully decoded and then re-encoded into AMR's narrowband codec — this is a transcode, not a remux. AMR Narrowband operates at a fixed sample rate of 8000 Hz and a single mono channel, so any stereo audio in the AVI will be downmixed and the sample rate will be resampled to 8 kHz. The result is a highly compressed, speech-optimized audio file suited for voice recordings rather than music.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles the decoding of the AVI container and audio stream, format conversion, and encoding to AMR. |
-i input.avi
|
Specifies the input file — an AVI container which typically holds a video stream (commonly H.264 or MJPEG) and an audio stream (commonly MP3 via libmp3lame). FFmpeg reads and demuxes both streams from this file. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the video stream from the AVI file. Since AMR is a pure audio format with no video container capability, this flag is required to produce a valid output. |
-c:a libopencore_amrnb
|
Selects the libopencore_amrnb encoder to transcode the audio into AMR Narrowband format. This fully re-encodes the audio (decoding the source codec such as MP3 first, then encoding to AMR-NB at 8000 Hz mono), rather than copying the stream. |
-b:a 12200
|
Sets the AMR bitrate mode to 12.2 kbps, which is the highest quality mode available in the AMR Narrowband standard and corresponds to the codec's Mode 7. This value must be one of eight predefined AMR bitrate integers; arbitrary values are not accepted by the libopencore_amrnb encoder. |
output.amr
|
Defines the output filename with the .amr extension, which tells FFmpeg to write the encoded audio using the IETF AMR single-channel file format — a lightweight container used for storing AMR Narrowband audio compatible with mobile devices and telephony systems. |
Common Use Cases
- Converting recorded interviews or meetings captured in legacy AVI format into AMR files for archiving in mobile-compatible voice systems
- Extracting dialogue or voiceover tracks from old AVI video productions to send via platforms that accept AMR voice messages, such as older mobile MMS systems
- Repurposing AVI-recorded field audio from digital camcorders into AMR format for integration with telephony or IVR (Interactive Voice Response) systems
- Stripping audio from AVI surveillance or dictation recordings to produce compact AMR files for long-term voice archive storage where file size is critical
- Preparing speech samples from AVI training videos for use in AMR-based voice synthesis or speech recognition pipeline testing
Frequently Asked Questions
Yes — this conversion involves noticeable quality loss, especially for anything beyond simple speech. AMR Narrowband (libopencore_amrnb) resamples audio to 8000 Hz and encodes in mono, which means stereo separation, music, and high-frequency content present in the AVI's audio track will be lost or heavily degraded. AMR was engineered specifically for human speech intelligibility on mobile networks, not for fidelity. If the AVI contains music or complex audio, AMR is the wrong target format.
AMR Narrowband (libopencore_amrnb) is hardcoded to operate at 8000 Hz sample rate with a single mono channel — these are not configurable parameters but fundamental properties of the codec standard. FFmpeg automatically downmixes stereo to mono and resamples the audio to 8 kHz when encoding to libopencore_amrnb. If you need stereo output, AMR is not an appropriate target format.
These values are the AMR codec's fixed bitrate modes, expressed in bits per second, defined by the ETSI AMR standard. The default of 12200 bps (12.2 kbps) is the highest quality mode and corresponds to the AMR 12.2 kbps mode widely used in 3GPP voice calls. Lower modes like 4750 bps offer extreme compression at the cost of speech intelligibility. Unlike typical audio codecs, AMR does not accept arbitrary bitrates — you must choose one of these eight predefined values.
Replace the value after -b:a in the command with any of the supported AMR bitrate modes: 4750, 5150, 5900, 6700, 7400, 7950, 10200, or 12200 (all in bits per second). For example, to use the lowest bandwidth mode you would write: ffmpeg -i input.avi -vn -c:a libopencore_amrnb -b:a 4750 output.amr. The default of 12200 provides the best speech quality within AMR Narrowband's range. Note that these are fixed codec modes, not freely adjustable bitrates.
Yes, on the command line you can use a shell loop to process multiple files. On Linux or macOS: for f in *.avi; do ffmpeg -i "$f" -vn -c:a libopencore_amrnb -b:a 12200 "${f%.avi}.amr"; done. On Windows Command Prompt: for %f in (*.avi) do ffmpeg -i "%f" -vn -c:a libopencore_amrnb -b:a 12200 "%~nf.amr". The browser-based tool processes one file at a time, but the displayed FFmpeg command is designed for easy adaptation to local batch workflows.
Largely no. AVI files can carry metadata tags in their INFO chunk (such as title, author, or creation date), but the AMR file format has no standardized metadata container to store this information. FFmpeg will not carry over AVI metadata fields to the AMR output. If metadata preservation matters for your workflow, consider extracting the audio to a format with richer metadata support such as M4A or OGG before converting.
Technical Notes
The libopencore_amrnb encoder used in this conversion is the open-source implementation of the 3GPP AMR-NB (Adaptive Multi-Rate Narrowband) codec, which operates exclusively at 8000 Hz mono. This represents a dramatic reduction in audio bandwidth compared to typical AVI audio tracks, which are usually MP3-encoded at 44100 Hz stereo. FFmpeg handles the necessary sample rate conversion and channel downmix automatically, but the resulting file will reflect AMR's inherent ceiling of approximately 3.4 kHz audio bandwidth — sufficient for voice but unsuitable for music or broadband audio. The AMR Narrowband codec uses one of eight fixed bitrate modes rather than a continuous bitrate scale; the default 12200 bps mode corresponds to the G.722.2 Annex A mode and delivers the best quality within the standard. The output .amr file uses the IETF AMR single-channel storage format with the '#!AMR
' magic header, which is compatible with most mobile devices and telephony platforms. Note that libopencore_amrwb (AMR Wideband) is also available and operates at 16000 Hz for higher speech quality, though this is not the default for AMR output. AVI files with multiple audio tracks will only have the first (default) audio stream extracted, as AMR does not support multiple tracks.
' magic header, which is compatible with most mobile devices and telephony platforms. Note that libopencore_amrwb (AMR Wideband) is also available and operates at 16000 Hz for higher speech quality, though this is not the default for AMR output. AVI files with multiple audio tracks will only have the first (default) audio stream extracted, as AMR does not support multiple tracks.