Convert FLV to AMR — Free Online Tool
Convert FLV Flash Video files to AMR audio, extracting and re-encoding the audio stream using the libopencore_amrnb codec optimized for speech. AMR's narrow-band compression makes it ideal for preserving voice content from FLV recordings at very low file sizes.
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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 reads the FLV container — which typically holds H.264 video and AAC or MP3 audio — and discards the video stream entirely, focusing only on the audio. The audio is then decoded and re-encoded from its original codec (usually AAC at 128k or higher) into AMR Narrow-Band format using the libopencore_amrnb encoder at 12,200 bps. AMR was engineered for speech telephony, so it uses a completely different compression model than AAC: rather than preserving a full frequency spectrum, it prioritizes the 300–3400 Hz range where human voice energy is concentrated. The result is a dramatically smaller file that sounds clear for speech but will noticeably degrade music or wide-spectrum audio.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In this browser-based tool, the equivalent FFmpeg.wasm binary runs locally in your browser via WebAssembly — no files leave your device. |
-i input.flv
|
Specifies the input FLV file. FFmpeg reads the FLV container, which may contain H.264 or Sorenson Spark video alongside AAC or MP3 audio — both streams are parsed, but only the audio will be used for AMR output. |
-c:a libopencore_amrnb
|
Sets the audio encoder to libopencore_amrnb, the open-source implementation of the AMR Narrow-Band codec. This re-encodes whatever audio codec was in the FLV (typically AAC) into the 3GPP AMR-NB format optimized for speech at 8 kHz. |
-b:a 12200
|
Selects the AMR-NB encoding mode at 12,200 bits per second, which is the highest quality mode available in the AMR Narrow-Band standard and corresponds to codec mode MR122. This gives the best speech intelligibility within the constraints of the AMR-NB format. |
output.amr
|
Defines the output filename with the .amr extension. FFmpeg uses this extension to write the file in the IETF AMR storage format (RFC 4867), which is directly playable on mobile phones and voice messaging systems that support AMR Narrow-Band. |
Common Use Cases
- Extracting spoken commentary or narration audio from an FLV screen recording to send as a voice message on a mobile platform that accepts AMR files
- Converting archived FLV lecture recordings or webinar captures into AMR format for playback on older mobile phones that support AMR natively
- Pulling voice-over audio from FLV promotional videos to store as compact AMR files for use in telephony IVR systems
- Reducing the file size of FLV interview or podcast content to AMR for transmission over low-bandwidth mobile connections
- Converting FLV news clip audio to AMR for integration into mobile journalism workflows where AMR is the native voice recording format
- Archiving the spoken audio track from historical FLV web content into a compact, telephony-standard format for long-term low-storage preservation
Frequently Asked Questions
Not particularly — AMR Narrow-Band is specifically engineered for the human voice frequency range (roughly 300–3400 Hz) and uses speech-optimized encoding algorithms. Music, sound effects, and ambient audio that contain frequencies outside this range will sound muffled, tinny, or distorted. AMR is best suited for FLV files whose primary audio content is spoken word, such as lectures, interviews, or voice-overs.
Two things happen simultaneously: the video stream is completely dropped (which alone removes the majority of an FLV file's size), and the remaining audio is encoded at only 12,200 bps using AMR's highly efficient speech compression. A 100MB FLV containing video and AAC audio might produce an AMR file of just a few hundred kilobytes. This extreme compression is by design — AMR was standardized for mobile telephony where bandwidth and storage are constrained.
AMR Narrow-Band (NB) operates at 8 kHz sample rate with bitrates from 4,750 to 12,200 bps, while AMR Wide-Band (WB) operates at 16 kHz with bitrates from 6,600 to 23,850 bps. Wide-Band produces noticeably better audio quality, especially for speech intelligibility. If your FLV contains clear speech and your target device supports AMR-WB (most modern smartphones do), you can replace libopencore_amrnb with libopencore_amrwb in the FFmpeg command for improved results. The default here uses NB for maximum device compatibility.
Replace the 12200 value after -b:a with any valid AMR Narrow-Band bitrate: 4750, 5150, 5900, 6700, 7400, 7950, 10200, or 12200 (bits per second). These are fixed mode bitrates defined by the AMR standard — values outside this set are not valid. For example, use -b:a 7950 for a slightly smaller file or keep -b:a 12200 for the highest quality AMR-NB output. Note that even 12,200 bps is far lower than typical AAC audio in FLV files, so some quality loss is inherent to this conversion.
Yes. On Linux or macOS, you can run a shell loop: for f in *.flv; do ffmpeg -i "$f" -c:a libopencore_amrnb -b:a 12200 "${f%.flv}.amr"; done. On Windows Command Prompt, use: for %f in (*.flv) do ffmpeg -i "%f" -c:a libopencore_amrnb -b:a 12200 "%~nf.amr". The browser-based tool processes one file at a time, so the FFmpeg command is especially valuable for bulk conversions of large FLV archives.
FLV files can carry metadata in their onMetaData event block (title, author, duration, etc.), but AMR is a bare audio format with essentially no standardized metadata container support. FFmpeg will attempt to map basic tags, but most FLV metadata will be silently dropped during this conversion. If metadata preservation matters, consider extracting the audio to a format like M4A or MP3 that has robust tagging support before converting to AMR.
Technical Notes
AMR Narrow-Band encoded with libopencore_amrnb enforces a strict 8,000 Hz sample rate and mono channel output — if your FLV contains stereo audio, FFmpeg will automatically downmix it to mono during encoding, and any stereo width or panning information will be lost. The libopencore_amrnb library implements the 3GPP AMR-NB standard (TS 26.071) and requires that bitrates match one of the eight defined codec modes exactly. FFmpeg's -b:a parameter in this context selects the closest valid AMR mode rather than performing free-form bitrate targeting as it would for MP3 or AAC. The output .amr file uses the IETF file storage format defined in RFC 4867, which adds a simple magic number header (#!AMR) and is recognized by most mobile platforms. One important limitation: because AMR-NB caps at 8 kHz sample rate, audio fidelity is equivalent to a standard telephone call — acceptable for voice intelligibility but unsuitable for any audio where tonal accuracy matters. There is no way to losslessly round-trip back to the original AAC or MP3 audio from the FLV; this conversion is strictly one-way and lossy.