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AMR (Adaptive Multi-Rate) is the voice codec designed for GSM cellular networks, operating at extremely low bitrates between 4.75 and 12.2 kbps. It was built for one purpose: encoding human speech efficiently enough to transmit over 2G mobile networks with limited bandwidth. Converting MP3 to AMR is a specialized operation typically needed for telephony systems, feature phone compatibility, or speech processing pipelines that require AMR input.
This conversion involves a dramatic quality reduction. MP3 files typically encode at 128-320 kbps across the full audible frequency range (20 Hz to 20 kHz), while AMR-NB operates at 8 kHz sample rate with a 300-3400 Hz bandwidth. The result sounds like a telephone call — adequate for intelligible speech but completely unsuitable for music. This conversion only makes sense when the target system specifically requires AMR format.
AMR remains the standard voice codec for GSM networks, VoIP telephony systems, and Interactive Voice Response (IVR) platforms worldwide. If you need to deploy pre-recorded audio prompts for a phone system, the prompts must typically be in AMR format. Similarly, feature phones from Nokia, Samsung, and Motorola that predate the smartphone era use AMR as their native voice recording and ringtone format.
Speech recognition engines and voice analysis tools sometimes require AMR input specifically. Telecoms testing equipment, call center quality monitoring systems, and cellular network analysis tools process AMR files because that is the format voice calls use over the air. Converting your MP3 voice recordings to AMR feeds them into these specialized systems without format compatibility issues.
FFmpeg decodes the MP3 source and re-encodes using the libopencore-amrnb encoder at 8 kHz mono with the AMR-NB codec. Output bitrate is selectable from 4.75, 5.15, 5.90, 6.70, 7.40, 7.95, 10.2, or 12.2 kbps — the standard AMR-NB mode set. The output file uses the simple AMR file format with a #!AMR\n magic header. Stereo MP3 sources are downmixed to mono since AMR only supports single-channel audio.
Extreme quality reduction is expected. AMR-NB at 12.2 kbps captures only the 300-3400 Hz frequency band at 8 kHz sample rate, producing telephony-grade audio. All bass below 300 Hz and all treble above 3400 Hz is discarded. Music converted through this path will be unrecognizable. Speech remains intelligible but gains the characteristic narrow, filtered sound of a phone call. This conversion should only be used when the target system requires AMR.
| Device | MP3 | AMR |
|---|---|---|
| Windows PC | Native | Partial |
| macOS | Partial | Partial |
| iPhone/iPad | Partial | Partial |
| Android | Native | Partial |
| Linux | Partial | Partial |
| Web Browser | Native |
Resolution: N/A
Bitrate: 320 kbps
OGG Vorbis preferred
Resolution: N/A
Bitrate: 256 kbps
AAC format required
Resolution: N/A
Bitrate: 128 kbps
Lossless FLAC/WAV for best quality
Resolution: N/A
Bitrate: 128 kbps
MP3 mono for spoken word
MP3 to AMR is a niche conversion for telephony, IVR systems, and legacy mobile devices. The dramatic quality reduction makes it unsuitable for music or general listening, but it produces the ultra-compact files that specialized voice systems require.
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| No |
| Ominaisuus | MP3 | AMR |
|---|---|---|
| Koko nimi | MPEG Audio Layer III | Adaptive Multi-Rate Audio |
| Tiedostopääte | .mp3 | .amr |
| Paras käyttökohde | Universal compatibility | Speech optimized |