Abstract / truncated to 115 words (read the full abstract)

A wide range of techniques for coding a single-channel speech and audio signal has been developed over the last few decades. In addition to pure redundancy reduction, sophisticated source and receiver models have been considered for reducing the bit-rate. Traditionally, speech and audio coders are based on different principles and thus each of them offers certain advantages. With the advent of high capacity channels, networks, and storage systems, the bit-rate versus quality compromise will no longer be the major issue; instead, attributes like low-delay, scalability, computational complexity, and error concealments in packet-oriented networks are expected to be the major selling factors. Typical audio coders such as MP3 and AAC are based on subband or transform ... toggle 4 keywords

audio coding linear predictive coding signal processing speech coding.


Biswas, Arijit
Technische Universiteit Eindhoven
Publication Year
Upload Date
Feb. 7, 2009

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