562results about "Source coding adaptation" patented technology

A bandwidth adjustment server (BWAS) (109) is provided which monitors system bandwidth usage, sends requests to terminals (102A, 102B and/or 106), to identify their coding capabilities, and directs each of the terminals to adjust their coding algorithms based on system bandwidth usage. If system bandwidth usage is high, the BWAS (109) requires the terminals to employ a less bandwidth intensive coding algorithm; similarly, when system bandwidth usage is low, the BWAS (109) will allow the terminals to employ higher bandwidth use coding algorithms. Codec renegotiation may be initiated if there is a disparity between the bandwidth allocated to new connections versus ongoing connections or an increase in data traffic.

A medium streaming distribution system reduces effects of packet loss in a network before the packet reaches radio base station. A medium distribution device for packet-transmits via the base station, a medium stream to the network by a real time transmission protocol. A packet analyzer monitors the packet arriving at the radio base station and transmits feedback Information associated with loss of a packet to the medium distribution device. Based on the feedback from a relay device and a terminal device of the medium stream, the transmission rates from the medium distribution device to the relay device and from the relay device to the terminal device are obtained to provide a greater transmission rate in a surplus band for re-transmission or a forward error correction.

Methods and apparatuses for improved data transmission control protocols. Acknowledgment streams are used by a sender process to generate various estimates of eligible transmission rates that are fair to other processes competing for bandwidth across a common network. In addition, the acknowledgment streams are used in conjunction with various other standard transmission control protocol metrics to adapt a filter for use on the rate estimates. Additional improvements in throughput may no be had by measuring the overall capacity of the computer network using packet pair dispersion measurements and adaptive network probing. In addition, the methods may be adapted for both packet transmission and video streaming applications.

A method for controlling real time data communications over a packet based network comprises steps of dynamically measuring data packet loss between a receiver and a sender on the network, dynamically measuring data latency between the sender and receiver, and comparing the dynamically measured data packet loss to a loss limit and comparing the dynamically measured data latency to a latency limit. If one or both of the loss limit or the latency limit is exceeded, then the method may include reducing the rate of data transmission from the sender.

A method for communicating video data on a wireless channel in a packet-switched network includes the steps of operating at a wireless terminal a compression in packets on the video data during a video coding operation, detecting wireless channel conditions and adapting control parameters of the video coding operation to the detected wireless channel conditions. The compression operation is a robust header compression operation and the step of adapting control parameters of said video coding operation is performed in dependence of information about the wireless channel conditions detected on a feedback channel made available in a decompression step associated with the compression operation.

During connection set up between a mobile telephone and a base station, program data is acquired from the network for controlling its mode of operation. The mobile telephone and base station each have a memory for storing program data relating to one or more modes of operation. The base station transmits a signal indicative of a preferred mode of operation and the mobile responds with a signal indicative of whether its memory already contains program data for this mode of operation. If not, then the base station transmits the required program data. The base station and mobile telephone then operate in this preferred mode. If the mobile is not so capable (for example because it has insufficient memory), the process is repeated with different mode control programs until the most appropriate mutually compatible program is identified. If a call attempt is initiated before this process is complete, the process is curtailed, such that the base station and mobile telephone identify programs already stored in the memory, but make no attempt to load a new program, as this could delay call set-up.

All data symbols used in data transmission of a modulated signal are precoded by hopping between precoding matrices so that the precoding matrix used to precode each data symbol and the precoding matrices used to precode data symbols that are adjacent to the data symbol in the frequency domain and the time domain all differ. A modulated signal with such data symbols arranged therein is transmitted.

A wireless network includes a plurality of proxy servers located at various locations to selectively transform data based upon network conditions, such as link congestion. A queueing model for the network can be used to determine optimal operating parameters for the network.

A coding system and method for a terminal including a multi-rate codec is disclosed. The terminal includes a multi-rate adaptive coder that is capable of transmitting a continuous voice stream transmission at a source code bit rate and a channel code bit rate. A quality of service probing module probes an end-to-end network path of the continuous voice stream transmission to obtain at least one quality of service parameter. A quality of service management module determines at least one constraint associated with the continuous voice stream transmission. An adaptive bit rate algorithm module dynamically adjusts the source code bit rate and the channel code bit rate as a function of the quality of service parameter and the constraint to obtain a maximum value of perceived user performance during the continuous voice stream transmission.

A data communication network for providing dynamic routing through both wireless and wired subnetworks to support wireless communication devices and wired remote stations is disclosed. In the wireless network, the wireless communication devices can be mobile RF terminals, while the wired remote stations might be personal computers attached to a wired subnet, such as an ethernet coaxial cable. The wireless network architecture utilizes a spanning tree configuration which provides for transparent bridging between wired subnets and the wireless subnets. The spanning tree configuration provides dynamic routing to and from wireless communication devices and remote stations attached to standard IEEE 802 LANs.

A “Media Transmission Optimizer” provides a media transmission optimization framework for lossy or bursty networks such as the Internet. This optimization framework provides a novel form of dynamic Forward Error Correction (FEC) that focuses on the perceived quality of a recovered media signal rather than on the absolute accuracy of the recovered media signal. In general, the Media Transmission Optimizer provides an encoder that optimizes the transmission of redundant frames of electronic media information encoded at different bit rates, and provides optimized playback quality by providing a decoder that automatically selects an optimal path through one or more available representations of each frame as a function of overall rate/distortion criteria.

A data transmitting method is disclosed. In transmitting A/V data stream in a transmitting device of a wireless network, the present invention includes transmitting the A/V data stream having a first data format to a receiving device, if a quality of a channel for carrying the A/V data stream to the receiving device is changed, deciding to change a data format of the A/V data stream to be transmitted to the receiving device into a second data format, and transmitting the A/V data stream having the second data format to the receiving device.

A method includes receiving a second audio frame at a decoder. The second audio frame follows a first audio frame in an audio signal and includes a first number of bits allocated to primary coding information associated with the second audio frame, a second number of bits allocated to redundant coding information associated with the first audio frame, and an indicator of a frame type of the first audio frame. In response to a frame erasure condition associated with the first audio frame, the second number of bits is determined based on the indicator and used to decode the first audio frame. In clean channel conditions, the first audio frame is received and decoded based on primary coding bits in the first audio frame, and the first number of bits is determined based on the indicator and used to decode the second audio frame.