Convert OGG to AMR — Free Online Tool
Convert OGG audio files (Vorbis or Opus streams) to AMR format using the libopencore_amrnb codec, optimized for speech encoding in mobile and telephony applications. This tool runs entirely in your browser — no uploads, no servers, just instant conversion powered by FFmpeg.wasm.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your OGG file here
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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
OGG is a container that typically carries wideband, high-fidelity audio encoded with Vorbis or Opus — codecs designed for music and general audio at moderate-to-high bitrates. AMR (Adaptive Multi-Rate) is an entirely different beast: a narrowband speech codec originally engineered for GSM mobile networks, operating at bitrates between 4.75 and 12.2 kbps. During this conversion, FFmpeg decodes the OGG audio stream (Vorbis or Opus) fully to raw PCM, then re-encodes it using libopencore_amrnb — the open-source AMR-NB encoder. Critically, AMR-NB is hardcoded to a single channel (mono) and an 8 kHz sample rate, so FFmpeg automatically downmixes stereo to mono and downsamples the audio. Any music or wideband content in the OGG file will sound noticeably degraded; AMR is purpose-built for speech intelligibility, not fidelity. Chapters and metadata tags present in the OGG file are not preserved, as the AMR container format does not support them.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool. In this browser-based tool, FFmpeg runs via WebAssembly (FFmpeg.wasm) entirely within your browser — the same command can be run locally on your desktop if you have FFmpeg installed. |
-i input.ogg
|
Specifies the input OGG file. FFmpeg will detect whether the OGG container holds a Vorbis, Opus, or FLAC audio stream and decode it accordingly to raw PCM before re-encoding to AMR. |
-c:a libopencore_amrnb
|
Sets the audio encoder to libopencore_amrnb, the open-source AMR Narrowband encoder. This codec enforces mono output at 8 kHz — appropriate for speech but not for music or wideband audio content from your OGG file. |
-b:a 12200
|
Sets the AMR-NB encoding bitrate to 12,200 bits per second (12.2 kbps), which is the highest available AMR-NB mode and produces the best speech intelligibility. Valid AMR-NB bitrates are discrete fixed values: 4750, 5150, 5900, 6700, 7400, 7950, 10200, and 12200 — arbitrary values between these are not valid. |
output.amr
|
Specifies the output filename with the .amr extension, which FFmpeg uses to confirm the AMR container format. The resulting file will be a raw AMR-NB bitstream compatible with mobile devices, telephony systems, and applications that support the AMR format. |
Common Use Cases
- Preparing voice recordings or dictation captured as OGG files for upload to telecom platforms or IVR systems that require AMR format
- Converting OGG podcast excerpts or speech segments to AMR for storage or playback on older feature phones and embedded devices with limited codec support
- Compressing OGG voice memos to extremely small file sizes (as low as ~4.75 kbps) for transmission over low-bandwidth or legacy mobile networks
- Transcoding OGG recordings of meetings or interviews to AMR to meet the input requirements of certain speech recognition or voice processing APIs
- Creating AMR audio assets for mobile applications that target legacy Android or Symbian devices where AMR is the native voice format
- Re-encoding OGG voice chat exports to AMR for archival in telephony systems that index and store calls in AMR format
Frequently Asked Questions
Yes, significantly — but the degree depends on your source content. AMR-NB is a narrowband codec engineered exclusively for speech: it operates at 8 kHz sample rate and mono only, which means any stereo music or wideband audio in your OGG file will be heavily downsampled and downmixed. If your OGG file contains voice recordings, the result will be intelligible and compact. If it contains music, the audio will sound thin, muffled, and noticeably degraded. This is an inherent limitation of the AMR format itself, not the conversion process.
No. AMR-NB (libopencore_amrnb) is strictly mono. When you convert a stereo OGG file to AMR, FFmpeg automatically downmixes the two channels into a single mono channel before encoding. If your OGG file already contains a single mono audio track — which is common for voice recordings — no channel information is lost during the downmix step.
They are lost. The AMR container format does not support chapters, embedded metadata tags, or multiple audio tracks — features that OGG natively provides. If you need to preserve metadata such as artist, title, or track information, you should store it externally before converting, as there is no way to embed it into the output AMR file.
Replace the value after -b:a with one of the valid AMR-NB bitrates: 4750, 5150, 5900, 6700, 7400, 7950, 10200, or 12200 (in bits per second). For example, to use the lowest bitrate for maximum compression, use -b:a 4750. The default in this tool is 12200, which is the highest AMR-NB bitrate and delivers the best speech intelligibility. Note that these are the only valid bitrate values for AMR-NB — arbitrary values are not accepted.
Yes. AMR-WB (Wideband) operates at 16 kHz sample rate instead of 8 kHz, which delivers significantly better speech quality. To use it, change -c:a libopencore_amrnb to -c:a libopencore_amrwb in the FFmpeg command, and update the -b:a value to a valid AMR-WB bitrate (e.g., 23850 for highest quality). The output file extension should remain .amr or you can use .awb. AMR-WB is supported on a wider range of modern devices and is a better choice when call quality is a priority over maximum compression.
The single-file command shown on this page must be adapted for batch use. On Linux or macOS, you can loop over files in a shell: for f in *.ogg; do ffmpeg -i "$f" -c:a libopencore_amrnb -b:a 12200 "${f%.ogg}.amr"; done. On Windows Command Prompt, use: for %f in (*.ogg) do ffmpeg -i "%f" -c:a libopencore_amrnb -b:a 12200 "%~nf.amr". This browser-based tool processes one file at a time, so the FFmpeg command is especially useful when you need to convert large batches or files over 1GB.
Technical Notes
The libopencore_amrnb encoder used in this conversion is an open-source implementation of the 3GPP AMR-NB standard (GSM 06.90), which defines eight fixed bitrate modes from 4.75 to 12.2 kbps. Because the codec is narrowband by design, FFmpeg will resample any audio above 8 kHz before encoding — this is automatic and non-optional. OGG files encoded with libopus often target 48 kHz sample rates and wide stereo fields, meaning the resampling and downmix applied during this conversion represent a dramatic reduction in audio bandwidth. OGG files encoded with FLAC inside the OGG container (OGG FLAC) are lossless at the source, but the AMR output will still be lossy regardless. The AMR file format itself has minimal container overhead — it is essentially a raw bitstream with a short magic number header — which is why it cannot carry metadata, chapters, or multiple audio tracks. If your use case involves mobile telephony integration or embedded system playback, confirm whether the target platform expects AMR-NB or AMR-WB, as they use different sample rates and are not interchangeable despite sharing the .amr extension in many implementations.