MyMediaTools

Convert AMR to OGG — Free Online Tool

Convert AMR audio files to OGG format using the Vorbis codec, transforming narrow-band mobile speech recordings into a widely supported open-format audio container. This is ideal for moving voice recordings off mobile devices into a format compatible with desktop players, web browsers, and open-source media ecosystems.

FFmpeg Command

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Estimated output:

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How It Works

AMR (Adaptive Multi-Rate) audio is decoded from its compressed speech-optimized bitstream — either AMR-NB (narrowband, 8 kHz sample rate) or AMR-WB (wideband, 16 kHz) — and then re-encoded into a Vorbis audio stream wrapped in an OGG container. Because AMR uses a completely different codec architecture designed specifically for telephony speech compression, this is a full transcode: the audio is fully decoded to PCM and then re-encoded with libvorbis. The output quality is governed by Vorbis's variable bitrate quality scale rather than AMR's fixed speech codec modes. Note that AMR's narrow 8 kHz or 16 kHz bandwidth is preserved in the output — the OGG file won't sound fuller than the original because the source audio simply doesn't contain higher-frequency content.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg command-line tool. In this browser-based tool, the same FFmpeg engine runs locally via WebAssembly (FFmpeg.wasm), so no files leave your device.
-i input.amr Specifies the input AMR file. FFmpeg will auto-detect whether it is AMR-NB or AMR-WB and select the appropriate decoder (libopencore_amrnb or libopencore_amrwb) to decode the telephony speech bitstream into raw PCM audio.
-c:a libvorbis Sets the audio encoder to libvorbis, which encodes the decoded AMR audio into a Vorbis stream. Vorbis is the default and most widely compatible audio codec for the OGG container, supported natively by all major browsers and most open-source media players.
-q:a 4 Sets the Vorbis variable bitrate quality level to 4 on a 0–10 scale, targeting approximately 128 kbps. For speech-bandwidth AMR source audio (typically 8–16 kHz), this quality level is more than sufficient and avoids encoding silence or speech artifacts at unnecessarily high bitrates.
output.ogg Defines the output filename with the .ogg extension, instructing FFmpeg to wrap the encoded Vorbis audio stream in an OGG container — an open, patent-free format developed by Xiph.Org that is well-suited for distributing and archiving voice recordings.

Common Use Cases

  • Converting voice memos or call recordings saved in AMR format on Android phones into OGG files for playback in desktop media players like VLC or foobar2000
  • Preparing speech recordings from mobile apps for upload to open-source podcast platforms or Funkwhale instances that prefer OGG/Vorbis over proprietary formats
  • Archiving WhatsApp or messaging app voice messages (which are often stored as AMR internally) into a more future-proof, openly documented container format
  • Transcribing mobile field recordings by converting AMR files to OGG so they can be imported into open-source audio editors like Audacity or Ardour
  • Standardizing a batch of AMR voice recordings from different devices into a single OGG format for consistent playback in a web-based audio player using the HTML5 audio element
  • Migrating speech data from a telephony system for use in a voice dataset or annotation pipeline that expects OGG/Vorbis files

Frequently Asked Questions

No — the audio quality ceiling is set by the original AMR recording. AMR-NB captures only up to 8 kHz of audio bandwidth, which is why it sounds narrow and telephony-like. AMR-WB reaches 16 kHz, which is noticeably clearer but still not full-range audio. The Vorbis encoder in the OGG file faithfully reproduces whatever was in the AMR source; it cannot reconstruct high frequencies that were never recorded. The conversion changes the codec and container, not the underlying audio content.
AMR is a proprietary codec tied to mobile telephony standards, and decoding it typically requires licensed or reverse-engineered libraries (like libopencore-amrnb). OGG is a fully open, patent-free container developed by Xiph.Org, and Vorbis is royalty-free with broad decoder support across operating systems, browsers, and media players. For archiving voice recordings, OGG/Vorbis is significantly more future-proof and doesn't depend on telephony-specific tooling to play back.
The quality is controlled by the -q:a flag, which sets Vorbis's variable bitrate quality level on a scale from 0 (lowest, roughly 64 kbps) to 10 (highest, roughly 500 kbps). The default used here is 4, which targets around 128 kbps — more than sufficient for speech-bandwidth AMR content. Since AMR-NB source audio is only 8 kHz, using a quality above 4 or 5 is largely wasteful for voice recordings; a value of 2 or 3 often produces indistinguishable results for speech. To use a lower quality, run: ffmpeg -i input.amr -c:a libvorbis -q:a 2 output.ogg
Yes. FFmpeg automatically detects whether the AMR file uses the narrowband (AMR-NB, decoded via libopencore_amrnb) or wideband (AMR-WB, decoded via libopencore_amrwb) codec and selects the appropriate decoder. The output command remains identical — FFmpeg handles the distinction transparently. AMR-WB files will produce noticeably cleaner OGG output since they contain audio up to 16 kHz rather than 8 kHz.
AMR files are extremely compact — AMR-NB operates at bitrates as low as 4.75 kbps, while the default Vorbis quality 4 targets roughly 128 kbps. This means the OGG output will typically be 5 to 20 times larger than the source AMR file. For a 1-minute AMR-NB voice message at 12.2 kbps, you might go from roughly 90 KB to over 900 KB in OGG. If file size is a concern, lower the -q:a value to 1 or 2, which still produces excellent quality for speech content.
AMR files have very limited metadata support — the format was designed purely for telephony transport, not media library management, so most AMR files carry no embedded tags. FFmpeg will pass through any metadata it can read from the AMR container into the OGG file's Vorbis comment tags, but in practice you will likely start with an empty tag set. If you want to add metadata to the output OGG file, you can append flags like -metadata title='My Recording' -metadata artist='Speaker Name' to the FFmpeg command before the output filename.

Technical Notes

The conversion from AMR to OGG involves two fundamentally different codec philosophies: AMR uses CELP (Code-Excited Linear Prediction) speech coding tailored for human voice at fixed low bitrates, while Vorbis uses a general-purpose psychoacoustic perceptual model suited to a wide range of audio content. Because of this, the transcode is lossy-to-lossy — audio quality can only be maintained or decreased, never improved. FFmpeg requires the libopencore-amrnb and libopencore-amrwb libraries to decode AMR; these are not always compiled into FFmpeg by default, so desktop users who build FFmpeg from source should confirm these libraries are enabled with --enable-libopencore-amrnb and --enable-libopencore-amrwb. The OGG container supports multiple audio tracks and chapter markers, but since AMR is a single-stream, mono or stereo format with no chapter support, none of those OGG container advantages are utilized in this conversion. AMR recordings are almost always mono (single channel); Vorbis will encode the output as mono accordingly, keeping file sizes down. For batch processing multiple AMR files in a shell, the command can be extended with a loop: for f in *.amr; do ffmpeg -i "$f" -c:a libvorbis -q:a 4 "${f%.amr}.ogg"; done

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