Conferencing systems are increasingly used for more elaborate presentations, often including music and sound effects. While speech remains the primary means for communication, content sharing is becoming more important and now includes presentation slides with embedded music and video files. In today’s multimedia presentations, playback of high-quality audio (and video) from DVDs and PCs is becoming a common practice; therefore, both the encoder and decoder must be able to handle this input, transmit the audio across the network, and play it back in sound quality that is true to the original.
New communications and telepresence systems provide High Definition (HD) video and audio quality to the user, and require a corresponding quality of media delivery to fully create the immersive experience. While most people focus on the improved video quality, telepresence experts and users point out that the superior audio is what makes the interaction smooth and natural. In fact, picture quality degradation has much lower impact on the user experience than degradation of the audio. Since telepresence rooms can seat several dozens of people, advanced fidelity and multichannel capabilities are required that allow users to acoustically locate the speaker in the remote room. Unlike conventional teleconference settings, even side conversations and noises have to be transmitted accurately to assure interactivity and a fully immersive experience.
Audio codecs for use in telecommunications face more severe constraints than general-purpose media codecs. Much of this comes from the need for standardized, interoperable algorithms that deliver high sound quality at low latency, while operating with low computational and memory loads to facilitate incorporation in communication devices that span the range from extremely portable, low-cost devices to high-end immersive room systems. In addition, they must have proven performance, and be supported by an international system that assures that they will continue to be openly available worldwide.
Audio codecs that are optimized for the special needs of telecommunications have traditionally been introduced and proven starting at the low end of the audio spectrum. However, as media demands increase in telecommunications, the International Telecommunication Union (ITU-T) has identified the need for a telecommunications codec that supports full human auditory bandwidth, that is, all sounds that a human can hear. This has led to the development and standardization of the G.719 audio codec...
The new white paper "G.719 - The First ITU-T Standard for Full-Band Audio" is available here: