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126results about How to "Packet loss" patented technology

System and method for real-time jitter control and packet-loss concealment in an audio signal

An “adaptive audio playback controller” operates by decoding and reading received packets of an audio signal into a signal buffer. Samples of the decoded audio signal are then played out of the signal buffer according to the needs of a player device. Jitter control and packet loss concealment are accomplished by continuously analyzing buffer content in real-time, and determining whether to provide unmodified playback from the buffer contents, whether to compress buffer content, stretch buffer content, or whether to provide for packet loss concealment for overly delayed or lost packets as a function of buffer content. Further, the adaptive audio playback controller also determines where to stretch or compress particular frames or signal segments in the signal buffer, and how much to stretch or compress such segments in order to optimize perceived playback quality.
Owner:MICROSOFT TECH LICENSING LLC

Data transport acceleration and management within a network communication system

Improved data transport and management within a network communication system may be achieved by utilizing a transmit timer incorporated within the sender device and exploiting host-level statistics for a plurality of connections between a sender and receiver. The period of the transmit timer may be periodically adjusted based on a ratio of the smoothed round-trip time and the smoothed congestion window, thereby reducing or eliminating bursty data transmission commonly associated with conventional TCP architectures. For applications having a plurality of connections between a sender and a receiver that share a common channel, such as web applications, the congestion window and smoothed round trip time estimates for all active connections may be used to initialize new connections and allocate bandwidth among existing connections. This aspect of the present invention may reduce the destructive interference that may occur as different connections compete with one another to maximize the bandwidth of each connection without regard to other connections serving the same application. Error recovery may also be improved by incorporating a short timer and a long timer that are configured to reduce the size of the congestion window and the corresponding transmission rate in response to a second packet loss with a predefined time period in order to increase resilience to random packet loss.
Owner:OPTIMORPHIX INC

System for handling variations in the reception of a speech signal consisting of packets

The present invention relates to a receiver system in a communication system supporting packet-based communication (e.g., an IP-network), including a receiver, speech decoder and a jitter buffer for handling delay variations in the reception of a speech signal consisting of packets containing frames with encoded speech. A jitter buffer controller is provided for keeping information about the functional size of the jitter buffer and for providing the speech decoder with control information, such that the speech decoder, based on that information, provides a dynamic adaptation of the size of the jitter buffer using the received encoded, packetized speech signal. The invention also relates to a method of adapting the function size of the jitter buffer of a receiver system.
Owner:TELEFON AB LM ERICSSON (PUBL)

Audio receiver having adaptive buffer delay

InactiveUS20060092918A1Buffer delay be increaseIncreasing and decreasing buffer delayError preventionTransmission systemsInterval delaySelf adaptive
Generally speaking, there are provided systematic techniques for increasing and decreasing jitter buffer delay. The disclosed techniques typically utilize various combinations of: evaluating received data over a specified interval, increasing a recommended buffer delay if the interval delay exceeds a first threshold and decreasing the recommended buffer delay if the interval delay is less than a second threshold, causing the recommended buffer delay to decrease over time until an underflow condition is identified, and / or increasing the recommended buffer delay in response to identifying the underflow condition.
Owner:PIVOT VOIP

Vertical handover

A vertical handover decision to perform a seamless vertical handover of a wireless apparatus from a current access network to a target access network is established. When deciding to perform the vertical handover from the current access network to the target access network it can be considered whether an intermediate access network could be used. It can be decided that first a seamless vertical handover is performed from the current access network to an intermediate access network, and subsequently a seamless vertical handover from the intermediate access network to the target access network is performed.
Owner:NOKIA TECHNOLOGLES OY

Method for encoding signals, related systems and program product therefor

A method for encoding video signals subjects the signals to unbalanced multiple description coding. The unbalanced multiple description coding codes a video signal in a first high resolution packet and a second low resolution packet and represents, respectively a first high resolution description and a second low resolution description. The unbalanced multiple description coding step includes using different intra refresh periods for the first and second high resolution descriptions, with an intra refresh period for the second low resolution description shorter than the intra refresh period of the first high resolution description.
Owner:STMICROELECTRONICS SRL
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