Convert AMR to OGA — Free Online Tool
Convert AMR audio files — the compressed speech format used in mobile voice recordings and telephony — into OGA files encoded with the Vorbis codec, producing open-format audio with significantly better fidelity and broader software compatibility. This is especially useful when you need to archive or edit voice recordings originally captured on a mobile device.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your AMR file here
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Settings
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
AMR (Adaptive Multi-Rate) stores audio using a narrow-band speech codec optimized for voice at extremely low bitrates, typically between 4.75 and 12.2 kbps. During this conversion, FFmpeg decodes the AMR stream using the libopencore_amrnb decoder, producing raw PCM audio, which is then re-encoded from scratch using the libvorbis encoder into an Ogg container with a .oga extension. Because AMR and Vorbis use completely different encoding algorithms — AMR is a speech-specific lossy codec while Vorbis is a general-purpose perceptual audio codec — a full transcode is required; there is no remuxing shortcut. The default Vorbis quality setting of -q:a 4 targets approximately 128 kbps variable bitrate, which is dramatically higher than a typical AMR file and produces noticeably cleaner audio, though some fidelity lost during the original AMR encoding cannot be recovered.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg media processing tool. In this browser-based tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) — no data leaves your device. |
-i input.amr
|
Specifies the input AMR file. FFmpeg detects the AMR container and selects the libopencore_amrnb decoder to read the compressed speech audio stream. |
-c:a libvorbis
|
Instructs FFmpeg to re-encode the audio using the libvorbis encoder, producing an Ogg Vorbis audio stream. This replaces AMR's narrow-band speech codec with a general-purpose perceptual codec suitable for archival and broad software playback. |
-q:a 4
|
Sets the Vorbis variable bitrate quality to level 4 on a 0–10 scale, targeting approximately 128 kbps. This is a substantial step up from a typical AMR file's 4.75–12.2 kbps and produces noticeably cleaner audio with fewer compression artifacts. |
output.oga
|
Defines the output filename with the .oga extension, which tells FFmpeg to write an audio-only Ogg container. The .oga extension is the correct designation for Ogg files holding audio-only streams such as Vorbis or FLAC. |
Common Use Cases
- Importing mobile voice memos or WhatsApp voice messages into a video or podcast editor that doesn't support AMR but handles OGA/Vorbis natively
- Archiving voice recordings from older Nokia, Samsung, or feature phones in a non-proprietary open format that will remain playable without vendor-specific software
- Preparing recorded speech or interview audio captured on a mobile handset for upload to platforms like SoundCloud or Bandcamp, which accept Ogg Vorbis but not AMR
- Converting bulk voicemail exports or call recordings from a VoIP system that outputs AMR into OGA for storage in a media library organized by open standards
- Improving perceived audio quality of a speech recording for a presentation or transcription service by transcoding from AMR's narrow-band 8 kHz limit into a Vorbis stream that a player can render more cleanly
Frequently Asked Questions
The conversion will not recover audio detail that AMR discarded when the original recording was made — that information is permanently gone. What it does do is re-encode the audio at a much higher bitrate using Vorbis's general-purpose perceptual codec, which eliminates the harsh codec artifacts that AMR introduces on non-speech sounds and makes the file more pleasant for playback in modern software. Think of it as moving the audio into a cleaner container with less aggressive compression, not restoring lost fidelity.
AMR was specifically engineered for mobile telephony at bitrates as low as 4.75 kbps, making it one of the most aggressive audio compression formats in common use. The default Vorbis quality level (-q:a 4) targets around 128 kbps variable bitrate, which is roughly 10–25 times higher. The larger file size reflects the more generous encoding budget Vorbis uses to represent the audio, even though the underlying speech content is the same.
Adjust the -q:a value in the command. The Vorbis quality scale runs from 0 (lowest, approximately 64 kbps) to 10 (highest, approximately 500 kbps), with 4 as the default targeting roughly 128 kbps. For voice recordings, -q:a 2 or -q:a 3 produces very good speech quality at a smaller file size, since Vorbis handles speech efficiently even at lower quality levels. Use a command like: ffmpeg -i input.amr -c:a libvorbis -q:a 2 output.oga
Yes, OGA supports FLAC as a lossless codec. You can run ffmpeg -i input.amr -c:a flac output.oga to produce a lossless Vorbis-alternative encoding inside the same Ogg container. However, since AMR is itself a lossy codec, encoding the output as FLAC provides lossless preservation of the already-lossy decoded PCM — it guarantees bit-perfect round-trip accuracy from this point forward but cannot undo AMR's original compression. The resulting file will be considerably larger than a Vorbis-encoded OGA.
AMR files have very limited metadata support — most contain no embedded tags at all beyond what the device firmware may optionally write. FFmpeg will attempt to copy any available metadata into the Ogg container's comment header, but in practice most AMR voice recordings arrive with no meaningful tags to transfer. If you need to embed title, date, or artist information in the output OGA file, you can add FFmpeg metadata flags to the command, for example: -metadata title='Interview 2024-01-15'.
Yes. On Linux or macOS, you can loop over files in a directory using a shell one-liner: for f in *.amr; do ffmpeg -i "$f" -c:a libvorbis -q:a 4 "${f%.amr}.oga"; done. On Windows with PowerShell, use: Get-ChildItem *.amr | ForEach-Object { ffmpeg -i $_.FullName -c:a libvorbis -q:a 4 ($_.BaseName + '.oga') }. The browser-based tool on this page processes one file at a time, so the FFmpeg command is the recommended approach for converting large batches of mobile voice recordings.
Technical Notes
AMR-NB (Narrowband), the default codec in AMR files, operates at a fixed 8 kHz sample rate — half the 16 kHz of AMR-WB (Wideband) — which means the audio contains no frequency content above 4 kHz. This is intentional for telephony but makes converted audio sound muffled compared to full-bandwidth recordings. When FFmpeg decodes the AMR stream and re-encodes it with libvorbis, the output sample rate defaults to 8000 Hz for AMR-NB sources; if you want the OGA file to be stored at a standard rate like 44100 Hz or 48000 Hz (for compatibility with players that expect it), add -ar 44100 to the command. The Ogg container used by OGA does support chapter markers, but AMR files carry no chapter data to transfer. Vorbis is not natively supported on Apple platforms without third-party plugins, so if playback on iOS or macOS is a priority, consider targeting libopus instead (-c:a libopus -b:a 64k), which is also supported in the OGA container and has broader hardware decoder support in modern environments.