Synchronized jitter buffers to handle codec switches

Abstract

Techniques are described for managing synchronized jitter buffers for streaming data (e.g., for real-time audio and / or video communications). A separate jitter buffer can be maintained for each codec. For example, as data is received in network packets, the data is added to the jitter buffer corresponding to the codec that is associated with the received data. When data needs to be read, the same amount of data is read from each of the jitter buffers. In other words, at each instance where data needs to be obtained (e.g., for decoding and playback), the same amount of data is obtained from each of the jitter buffers. In addition, the multiple jitter buffers use the same playout timestamp that is synchronized across the multiple of jitter buffers.

Description

Background technique [0001] Real-time communication services are used to communicate audio, video, and other types of streaming data over computer networks, such as the Internet. To provide a good user experience, real-time communication services need to deliver streaming data with low latency, while being able to handle changing network conditions (eg, changes in bandwidth and reliability). [0002] In some solutions, a single codec is used for a real-time communication session (eg, for an audio call). A single codec can be associated with a single jitter buffer that manages session latency. However, the ability of a single codec and associated jitter buffer to adapt to different network conditions is limited. [0003] In some solutions, multiple codecs and associated jitter buffers are used for a real-time communication session. While such solutions may allow codecs to be switched during a communication session, they suffer from a number of problems. For example, such so...

AI Technical Summary

Problems solved by technology

For example, such solutions may lose data during codec switching and / or may not handle frequent codec switching

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