Network audio transmission is becoming increasingly popular within the broadcast community, with applications to Voice over IP (VoIP) communications, audio content distribution, and radio broadcast. Issues of end-to-end latency, jitter, and overall quality, including glitches of the delivered signal, all impact on the value of the technology. Although considerable literature exists comparing audio codecs, little has been published to compare systems in terms of their real-word performance. In response, we describe methods for accurately assessing the quality of audio streams transmitted over networks. These methods are then applied to an empirical evaluation of several audio compression formats supported by different streaming engines.
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