At+t Corp.

Robert Patrick Lyons

Howard Paul Katseff

Bethany Scott Robinson

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.

Low-latency audio interface for packet telephony

In a method for reducing latency in packet telephony caused by anti-jitter buffering, audio data elements are received and placed in a telephony input buffer used for anti-jitter buffering. Rather than wait until the buffer is full, the audio data elements are clocked, or played, out of the buffer at a rate slower than the normal play rate. In this way, latency due to the initial buffer fill period is reduced or eliminated. Audio data elements continue to be played out at a slower than normal rate until the buffer fill level reaches a threshold. At that time, the play rate for sending data elements out of the telephony input buffer is adjusted to the normal play rate. In an alternative embodiment of the present invention, the fill level of the telephony input buffer is controlled within a desired range by speeding up or slowing down the rate at which audio data elements are played out of the telephony input buffer. In yet another alternative embodiment, the amount of latency jitter in the packet network is measured and the size of the telephony input buffer is adjusted based upon the relative amount of jitter, such that the relative size of the buffer is reduced when the packet network is quiet, and the size of the buffer is increased when the network is relatively jittery.

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Jackson, NJ, United States of America

Robert Patrick Lyons