Abstract: Apparatus for controlling at least one variable output parameter in response to a variable predetermined input parameter in a process system, such as a digital thermostat. The apparatus provides adaptive control of the output variable by utilizing a controller means that includes an adaptive controller means, an identifier means and a tuner means. The identifier means defines a model having parameters which represent the operational characteristics of the process system, and the identifier means monitors the operation of the adaptive controller means and selectively changes the parameters of said model to improve the operation of the adaptive controller means. The tuner means receives the model parameters from the identifier means and calculating robust and reliable values of said predetermined gain factors and applying the same to the adaptive controller means for use thereby.

Abstract: A computationally simple yet powerful forward error correction code scheme for transmission of real-time media signals, such as digitized voice, video or audio, in a packet switched network, such as the Internet. For each window of k data packets, the invention generates and transmits at least one cross-wise parity packet taken as an index-shifted function over the k data packets. The invention thereby enables a receiving end to recover from packet loss.

Abstract: A computationally simple yet powerful forward error correction code scheme for transmission of real-time media signals, such as digitized voice, video or audio, in a packet switched network, such as the Internet. The invention appends to each of a series of payload packets a single forward error correction code that is defined by taking the XOR sum of a preceding specified number of payload packets. The invention thereby enables correction from the loss of multiple packets in a row, without significantly increasing the data rate or otherwise delaying transmission.

Abstract: A method for improving perception of electronic content from a computer network such as the Internet or an intranet. Network latencies and the type of electronic content such as text, graphical images, animation, voice, video and other electronic content interact to influence user perception of the quality of information provided. As network latency increases and becomes more variable, users typically become less satisfied. The method dynamically adjusts the amount of electronic content presented to user based on a determined network latency. The amount of electronic content is also adjusted progressively and underlying transport protocol such as Transmission Control Protocol (“TCP”) and User Datagram Protocol (“UDP”) are adaptively adjusted based on the type of electronic content requested (e.g., TCP for text, UDP for graphical images, etc.).

Abstract: A method and apparatus for communicating a real time media input over a network. The apparatus encodes the input into data packets having a number of frames ordered according to a first variable. A receiving device unpacks and buffers the unpacked data packets for playout according to a second variable. The receiving device generates utility parameters for evaluating a dynamic characteristic of the network that transports the data packets. The receiving device selects a preferred utility parameter and adjusts the first and second variable according to the selected utility parameter. The method includes encoding an analog input into data packets that are transported to a receiving device. The method also includes unpacking the data packets, buffering the unpacked data packets according to a second variable, and generating at least two utility parameters that represent a dynamic characteristic of a network.

Abstract: A method and apparatus for improving the speed and quality of end-to-end data or real-time media transmissions over an internet is disclosed. A media stream being transmitted to the internet is channel coded at the edge of the internet in order to free upstream bit rate for use in source coding the media. The channel coded media stream may then be decoded at a remote edge of the internet to recover lost packets.

Abstract: A method and apparatus for improving the speed and quality of end-to-end data or real-time media transmissions over an internet is disclosed. A data stream representing a media signal at a given level of compression is processed just before the data stream enters the internet. A less compressed data stream representing the same media signal is generated transmitted through the internet. Due to the lower level of compression, the underlying media signal is less sensitive to packet loss in the internet and, as a result, the media signal that arrives at the receiving end will tend to be more continuous and clear.

Abstract: A computationally simple yet powerful forward error correction code scheme for transmission of real-time media signals, such as digitized voice, video or audio, in a packet switched network such as the Internet. An encoder at the sending end derives p redundancy blocks from each group of a k payload blocks and concatenates the redundancy blocks, respectively, with payload blocks in the next group of k payload blocks. At the receiving end, a decoder may recover up to p missing packets in a group of k packets, provided with the p redundancy blocks carried by the next group of k packets. The invention thereby enables correction from the loss of multiple packets in a row, without significantly increasing the data rate or otherwise delaying transmission.

Abstract: A method and apparatus are used in a gateway to discard selected frames received with a selected encoded-information-type from a communication link with a larger bandwidth to avoid overflowing an internal delay variance removing queue used for protocol translation to a communication link with a smaller bandwidth. The discarded frames do not decrease the quality of translated information. A visual delay variance removing queue congestion indicator is included to indicate three levels of congestion in the delay variance removing queue for received frames. The method and apparatus are used in a multimedia gateway which is translating audio/video conferencing protocols (e.g., H.320, H.323/LAN H.323/PPP and H.324) received from a communication link with a large bandwidth and sent to a communication link with a smaller bandwidth.

Abstract: An improved system for identifying the loudest speech signal in a G.723.1 based audio teleconferencing link is disclosed. The system selects the loudest of several analog audio signals by directly analyzing the encoded G.723.1 bit streams representing those signals, rather than by decoding the encoded speech signal in the G.723.1 bit streams and then re-encoding the signal as a selected output bit stream. The system uses the excitation gain parameters encoded in G.723.1 frames to approximate frame gains for respective bit streams and then estimates a short term speech energy for each bit stream by averaging the approximate frame gains over time. The system then compares the estimated speech energy levels and outputs to each conference participant the signal with the highest estimated speech energy as the next portion of an output signal.

Abstract: A computationally simple yet powerful forward error correction code scheme for transmission of real-time media signals, such as digitized voice, video or audio, in a packet switched network, such as the Internet. The invention appends to each of a series of payload packets a single forward error correction code that is defined by taking the XOR sum of a preceding specified number of payload packets. The invention thereby enables correction from the loss of multiple packets in a row, without significantly increasing the data rate or otherwise delaying transmission.

Abstract: The present invention provides a method (200, 400) and device (500) for, within a variable or fixed block size video compression scheme, providing optimal bit allocation among at least three critical types of data: segmentation, motion vectors and prediction error, or DFD. Since the amount of information represented by one bit for a particular type of data is not equivalent to the information represented by one bit for some other data type, this consideration is taken into account to efficiently encode the video sequence. Thus, a computationally efficient method is provided for optimally encoding a given frame of a video sequence wherein, for a given bit budget the proposed encoding scheme leads to the smallest possible distortion and vice versa, for a given distortion, the proposed encoding scheme leads to the smallest possible rate.

Abstract: A method (100, 200), device (300) and microprocessor (400) are provided for selectively compressing video frames of a motion compensated prediction-based video codec based on a predetermined set of compression techniques. An energy estimate of the current displaced frame difference, DFD, is used to compute a ratio between the estimate and a historical mean of energy estimates. The ratio is iteratively compared to a predetermined set of thresholds which are associated with the predetermined set of compression techniques. The comparisons are used to choose a technique based on the thresholds, and a technique is selected to be used for encoding the current DFD.