Audio format developed by the Nippon Telegraph and Telephone Corporation (NNT Labs); uses transform-domain weighted interleaved vector quantization to compress audio data; can be encoded in 80, 96, 112, 128, 160, and 192 kbps bitrates.

This format was originally intended for voice compression and is able to change the bit rates of encoded audio data. AMR is actively used for encoding and decoding voice while its being transmitted over network channels. The format allows one to consider network capacity and adapt it to channel loads. Due to this, the voice quality is practically undiminished.