System and method for improving audio quality on a conferencing network

Abstract

A conferencing system (,) improves audio quality by identifying and selectively allowing a dominant audio channel to bypass a lossy compression transformation included in a server,. The system (,) places conference calls over a network (,) and includes the conference server (,), having an input interface (,), a decompressor (,), a summation circuit (,), a correlation unit (,) a compressor (,), and an output interface (,). The input interface (,) receives a plurality of audio channels, each being compressed using a lossy compression algorithm. The decompressor (,) decompresses each of the incoming channels to produce a plurality of decompressed audio channels, which are then summed by the summation circuit (,) to produce a summation stream. The compressor (,) compresses the summation stream to produce a first audio output. The correlation unit (,) identifies a dominant channel from the incoming compressed audio channels. The dominant channel bypasses decompressor (,) and compressor (,) to be directly output as a second audio channel. The first and second output channels are then transmitted over the network (,) to a plurality of clients (,). The clients (,) present both channels of audio data to conference participants. Overall audio quality is improved because the dominant channel bypasses successive decompression/compression steps.