Convert OGA to AMR — Free Online Tool
Convert OGA audio files (Ogg container with Vorbis or FLAC audio) to AMR format using the libopencore_amrnb codec, optimized for speech encoding in mobile and telephony applications. This conversion trades the high-fidelity, open-source OGA format for AMR's ultra-compact, speech-tuned compression — ideal when compatibility with GSM networks or voice messaging systems is required.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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 decodes the audio stream from the OGA container — whether it was encoded with Vorbis, FLAC, or Opus — and re-encodes it entirely using the libopencore_amrnb codec into an AMR Narrowband file. AMR-NB operates at a fixed 8 kHz sample rate and is engineered around human speech frequencies, so FFmpeg will automatically resample your OGA audio (which may have been recorded at 44.1 kHz or 48 kHz) down to 8 kHz mono. This is a destructive, lossy transcoding process — the wide frequency range and stereo information present in the original OGA file cannot be preserved in AMR. The output bitrate defaults to 12,200 bps (AMR-NB's highest quality mode), which offers the best intelligibility possible within the format's constraints.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the underlying engine that this browser tool wraps via WebAssembly. The same command runs identically on desktop FFmpeg installations for files over 1 GB. |
-i input.oga
|
Specifies the input file in OGA format. FFmpeg will detect whether the audio stream inside is Vorbis, FLAC, or Opus and decode it accordingly before re-encoding to AMR. |
-c:a libopencore_amrnb
|
Selects the libopencore_amrnb encoder, which implements the AMR Narrowband codec. This encoder forces the output to 8 kHz mono, regardless of the sample rate or channel count of the source OGA file. |
-b:a 12200
|
Sets the AMR-NB bitrate to 12,200 bps, which is the highest of the eight fixed bitrate modes defined by the AMR-NB standard. This mode provides the best speech intelligibility the format can deliver and is the recommended default for converting from higher-quality OGA sources. |
output.amr
|
Defines the output filename with the .amr extension. FFmpeg uses this extension to select the AMR container format, which wraps the libopencore_amrnb-encoded frames in the standard 3GPP AMR file structure recognized by mobile devices and telephony software. |
Common Use Cases
- Preparing voice memos or spoken-word recordings stored in OGA format for upload to a telephony platform or IVR system that only accepts AMR files
- Converting FLAC-encoded OGA audio of a speech or interview to AMR for storage on older mobile devices or feature phones with limited codec support
- Archiving Vorbis-encoded OGA voice recordings into the AMR format required by certain 3GPP multimedia messaging (MMS) pipelines
- Reducing file size of OGA-encoded speech recordings for transmission over low-bandwidth mobile networks where AMR's compact bitrates are essential
- Converting OGA audio notes or dictations into AMR format compatible with legacy Android or Nokia handsets that natively decode AMR
- Producing AMR audio assets from OGA source files for integration into embedded systems or automotive infotainment units that use AMR for voice prompts
Frequently Asked Questions
Yes, and the degradation can be significant depending on your source material. OGA with Vorbis or FLAC audio typically supports full-fidelity stereo at 44.1–48 kHz, while AMR-NB is constrained to 8 kHz mono — roughly telephone quality. The conversion is optimized for speech intelligibility, not music or rich soundscapes. If your OGA file contains music, sound effects, or high-frequency content, that information will be lost entirely in the AMR output.
AMR Narrowband (libopencore_amrnb) is inherently a mono codec — it does not support stereo audio. FFmpeg automatically downmixes both channels of a stereo OGA file to a single mono channel during encoding. This is a hard limitation of the AMR-NB format specification, not a quirk of this tool or the FFmpeg command. If preserving stereo is critical, AMR is not the appropriate target format for your use case.
No. OGA supports metadata tags and even chapter markers, but the AMR container format has no standardized mechanism for storing this kind of metadata. FFmpeg will silently discard tags such as artist, title, album, and any chapter data during the conversion. If preserving metadata matters, you should keep the original OGA file alongside the AMR output.
Replace the value after -b:a with one of the eight bitrates AMR-NB supports: 4750, 5150, 5900, 6700, 7400, 7950, 10200, or 12200 (all in bits per second). For example, to use the lowest bitrate, the command becomes: ffmpeg -i input.oga -c:a libopencore_amrnb -b:a 4750 output.amr. Lower bitrates produce smaller files but reduced speech clarity, while 12200 (the default used here) gives the best quality AMR-NB can offer.
The single-file command shown on this page processes one file at a time, but you can adapt it for batch processing in a shell. On Linux or macOS, run: for f in *.oga; do ffmpeg -i "$f" -c:a libopencore_amrnb -b:a 12200 "${f%.oga}.amr"; done. On Windows Command Prompt, use: for %f in (*.oga) do ffmpeg -i "%f" -c:a libopencore_amrnb -b:a 12200 "%~nf.amr". This is especially useful for large batches that exceed the browser tool's 1 GB file limit.
AMR-NB (the default in this tool) uses an 8 kHz sample rate and suits standard telephony. AMR-WB (Wideband) uses 16 kHz and delivers noticeably better speech quality at the cost of slightly larger files. To use AMR-WB, change the codec flag in the command to -c:a libopencore_amrwb and use a compatible bitrate such as -b:a 23850. Choose AMR-WB if your target device or platform supports it and voice intelligibility is the priority.
Technical Notes
The libopencore_amrnb codec implements the 3GPP AMR-NB standard (TS 26.071), which was designed specifically for GSM and UMTS voice transmission. Because AMR-NB is hard-locked to 8 kHz / mono, FFmpeg must perform both sample rate conversion (downsampling from whatever rate the OGA source uses) and channel downmix before encoding — making this one of the more transformative conversions possible from the OGA format. OGA files encoded with FLAC (lossless) will suffer the most perceptible quality reduction, as lossless audio is compressed into one of eight fixed AMR bitrate modes, none of which exceed 12,200 bps. The AMR file format itself uses the .amr extension and a simple frame-based container with a magic header; it is not related to Ogg's streaming container model at all. One practical concern: if your OGA source was encoded with libopus at a wide bandwidth setting, the downsampling to 8 kHz will alias any content above 4 kHz, which could introduce ringing artifacts in the AMR output — using a pre-filter or choosing a higher AMR bitrate mode will not eliminate this, as the sample rate ceiling is absolute.