Audio - AMR

 

Adaptive Multi-Rate (AMR) is an Audio data compression scheme optimized for speech coding . AMR is adopted as the standard speech codec by 3GPP .

The codec has eight bit rates, 12.2, 10.2, 7.95, 7.40, 6.70, 5.90, 5.15 and 4.75 kbit/s. The bitstream is based on frames which contain 160 samples and are 20 milliseconds long. AMR uses different techniques, such as Algebraic Code Excited Linear Prediction (ACELP), Discontinuous Transmission (DTX), voice activity detection (VAD) and comfort noise generation (CNG).

The idea to use various modes arises from the network conditions. If the channel is bad, source coding is reduced and channel coding is increased. This improves the quality and robustness of the network. In the particular case of AMR this improvement is somewhere around 4-6 dB S/N for useable communication. The usage of AMR requires optimized link adaptation that selects the best codec mode to meet the local radio channel and capacity requirements. The new intelligent system allows the network operator to prioritize capacity or quality per base station.

Adaptive Multi Rate - WideBand or AMR-WB is a speech coding standard developed after the AMR using same technology like ACELP . The codec provides excellent speech quality due to wider speech bandwidth of 50 - 7000 Hz compared to narrowband speech codecs which in general are optimized for POTS wireline quality of 300-3400Hz. AMR-WB is codified as G.722.2 , an ITU-T standard speech codec.

AMR modes

AMR-WB operates like AMR with various bit rates. The bit rates are the following: 6.60, 8.85, 12.65, 14.25, 15.85, 18.25, 19.85, 23.05 and 23.85 kbit/s. The lowest bit rate providing excellent speech quality in a clean environtment is 12.65 kbit/s. Higher bit rates are useful in background noise conditions and in the case of music. Also lower bit rates of 6.60 and 8.85 provide reasonable quality especially if compared to narrow band codecs.

Future

AMR-WB is already standardized for future usage in networks such as UMTS . There it provides so much higher speech quality that it seems probable that also older networks will have to gradually be transformed to support wide band.

AMR-WB+ is an extension of AMR-WB . It adds support for stereo signals and higher sampling rates. Another main improvement is the use of transform coding additionally to ACELP . This greatly improves the generic audio coding. Automatic switching between transform coding and ACELP provides both good speech and audio quality with moderate bit rates.

As AMR-WB operates at internal sampling rate 12.8 kHz, AMR-WB+ also supports various internal sampling frequencies ranges from 12.8 kHz to 38.4 kHz.

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* The above text is quoted from Wikipedia

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