The Cisco TelePresence system codec uses advanced microphone, speaker, and audio encoding technologies to preserve the quality and directionality of the audio so that it appears to emanate from the location of the person speaking at the same volume as it would if that person were actually sitting across the table from you.
The quality of the audio enjoyed by the meeting participants is a function of three variables:
- Frequency spectrum captured by the microphones: The Cisco TelePresence microphones are designed to capture a 48-kHz spectrum of audio frequencies. The Cisco TelePresence speakers are designed to reproduce that same rich frequency spectrum.
- Spatiality (that is, directionality) of the audio: To preserve the spatiality (that is, directional perception) of the audio, for larger Cisco TelePresence systems, individual microphones are placed at specific locations on the virtual table, along with speakers located under each display.
- Degree of compression applied to the original audio signal: Inside the Cisco TelePresence Codec an onboard array of DSPs encode the audio signal from the microphones into RTP packets using the Advanced Audio Coding-Low Delay (AAC-LD) encoding and compression standard. The resulting bandwidth required to transport the audio signals between the systems is 64kbps per microphone. Therefore, a CTS-3X00 requires 192kbps of audio bandwidth, whereas the CTS-1000 or CTS-500 requires 64kbps of audio bandwidth.
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