| 摘要 |
In a method for reducing latency in packet telephony caused by buffering at the conversion stage between analog audio signals and digital audio data, analog audio is sampled at a rate far greater than necessary for telephony. The increased sampling rate allows the audio data to pass much more rapidly through the data conversion buffer. After passing through the buffer, the data is downsampled to a rate normally used for telephony. To handle audio data for speaker output, the data is upsampled to a rate far in excess of the rate necessary for processing telephony-grade voice signals. The increased sampling rate allows the audio data to pass much more rapidly through the data conversion buffer. After passing through the buffer, the data is converted into an analog audio signal for sending to the speaker. In this way, latency due to the buffering that accompanies the process of converting audio signals to digital data, or vice versa, is minimized.
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