| 摘要 |
A method and system to minimize the potential of jitter buffer underflow/overflow resulting from a difference in sampling rates of an audio encoder and an audio decoder are disclosed herein. The difference in sampling rates, or clock skew, can be determined from a difference between an actual amount of data stored in a jitter buffer and the desired, or threshold, amount. A subset of packets from a sequence of packets output to the audio decoder can be altered to compensate for the clock skew, whereby the amount of data associated with the subset of packets is decreased when the sampling rate of the encoder is greater than the sampling rate of the decoder, and the amount of data is increased when the sampling rate of the encoder is less than the sampling rate of the decoder. The present invention finds particular advantage in providing audio data via a packet-switched network.
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