Abstract: A packet scheduler reduces or “compresses” the packet transmission delay jitter or delay range where packets experience little or no scheduling delay before transmission. As a result, the number of packets that experience little or no delay is reduced. A preferred example way of compressing the packet transmission delay jitter is to reduce the transmission priority of low delay packets. Compressing the delay jitter is particularly desirable for services like VoIP that require low packet transmission delay jitter.

Abstract: Perceived interactivity in user communications is achieved by reducing a perceived delay switching the active transmitter in the communication without having to reduce actual transmission and setup delays associated with a communication exchange. A sound signal is identified in the user communication. The sound signal is analyzed to identify or estimate a sound signal segment. The sound signal segment is preferably (though not necessarily) located at the beginning or the end of the sound signal. The sound signal segment may be selected directly from the sound signal itself, from a modified version of the sound signal, or from a signal associated with the sound signal. A determination is made that a length or duration of the sound signal segment should be or can be modified. One or more modifications for the sound signal segment are determined and are provided to one or more processing units to perform the modification(s).

Abstract: Audio artifacts due to overrun or underrun in a playout buffer caused by the sampling rates at a sending and receiving side not being at the same rate are reduced. An LPC-residual is modified on a sample-by-sample basis. The LPC-residual block, which includes N samples, is converted to a block comprising N+1 or N?1 samples. A sample rate controller decides whether samples should be added to or removed from the LPC-residual. The exact position at which to add respective remove samples is either chosen arbitrarily or found by searching for low energy segments in the LPC-residual. A speech synthesiser module then reproduces the speech. By using the proposed sample rate conversion method the playout buffer can be continuously controlled. Furthermore, since the method works on a sample-by-sample basis the buffer can be kept to a minimum and hence no extra delay is introduced.

Abstract: Low rate data signals, such as standard TTY signals and standard DTMF signals, can be effectively analyzed and encoded by speech encoders employed in conventional digital cellular system. This is accomplished by providing a low rate data encoder in the transmission path of the low rate data signal, just prior to the speech encoder used by the digital cellular system. In so doing, the standard, low rate data signal is transformed (i.e., modified) by the low rate data encoder into a signal that is more compatible with the speech encoder. Similarly, a low rate data decoder is provided at the receiving end of the transmission path to decode the modified, low rate data signal.