I am wondering about the better audio compression technique that can be used in real time with a higher compression ratio, low computational complexity, and low latency.
Actually, I am confused between vector quantization and fractal coding. Which one can be implemented in real-time enviroment?
The Adaptive Multi-Rate (AMR or AMR-NB or GSM-AMR) audio codec is an audio compression format optimized for speech coding. AMR speech codec consists of a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates ranging from 4.75 to 12.2 kbit/s with toll quality speech starting at 7.4 kbit/s.
In general, audio quality, data rate, latency, and complexity are to a certain extent mutually exclusive requirements. Depending where you strike the balance, different techniques may prove best.
If you are interested in speech coding, a plethora of algorithms exist for varying requirements. AMR is one of them, the ITU G.7xx family also has a lot to offer. For more general audio coding, there are fewer established techniques with really low latency. Approaches based on ADPCM, possibly in subbands are probably most common.
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