7837 results about "Jitter" patented technology

A method and apparatus are provided that allows for the representation of a larger number of classes of network traffic and logical queues than is physically available on a per port basis within a network device. A number of logical queues, whose number can match the number of classes of network traffic a network device handles, may be supported across an aggregated set of ports even though the network device has fewer physical queues per port than there are classes of network traffic. The method and apparatus improve the management of network traffic sensitive to time delay and jitter, and further facilitates the operation of these applications in a simultaneous or near simultaneous manner.

Methods and systems are provided for allocating resources including VoIP (voice over Internet Protocol) and Non-VoIP resources. In some embodiments, multiplexing schemes are provided for use with OFDMA (orthogonal frequency division multiplexing access) systems, for example for use in transmitting VoIP traffic, in some embodiments, various HARQ (Hybrid Automatic request) techniques are provided for use with OFDMA systems. In various embodiments, there are provided methods and systems for dealing with issuea such as Handling non-full rate vocoder frames, VoIP packet jitter handling, VoIP capacity increasing schemes, persistent and non-persistent assignment of resources in OFDMA systems.

Embodiments of the invention enable minimum latency site independent real-time video transport over packet switched networks. Some examples of real-time video transport are video conferencing and real-time or live video streaming. In one embodiment of the invention, a network node transmits live or real-tine audio and video signals, encapsulated as Internet Protocol (IP) data packets, to one or more nodes on the Internet or other IP network. One embodiment of the invention enables a user to move to different nodes or move nodes to different locations thereby providing site independence. Site independence is achieved by measuring and accounting for the jitter and delay between a transmitter and receiver based on the particular path between the transmitter and receiver independent of site location. The transmitter inserts timestamps and sequence numbers into packets and then transmits them. A receiver uses these timestamps to recover the transmitter's clock. The receiver stores the packets in a buffer that orders them by sequence number. The packets stay in the buffer for a fixed latency to compensate for possible network jitter and/or packet reordering. The combination of timestamp packet-processing, remote clock recovery and synchronization, fixed-latency receiver buffering, and error correction mechanisms help to preserve the quality of the received video, despite the significant network impairments generally encountered throughout the Internet and wireless networks.

Packet-based central conference bridges, packet-based network interfaces and packet-based terminals are used for voice communications over a packet-based network. Modifications to these apparatuses can reduce the latency and the signal processing requirements while increasing the signal quality within a voice conference as well as point-to-point communications. For instance, by selecting the talkers prior to the decompression of the voice signals, decreases in the latency and increases in signal quality within the voice conference can result due to a possible removal of the decompression and subsequent compression operations in a conference bridge unnecessary in some circumstances. Further, the removal of the jitter buffers within the conference bridges and the moving of the mixing operation to the individual terminals and/or network interfaces are modifications that can cause lower latency and transcoding within the voice conference.

Embodiments of the invention enable the synchronization of clocks across packet switched networks, such as the Internet, sufficient to drive a jitter buffer and other quality-of-service related buffering. Packet time stamps referenced to a local clock create a phase offset signal. A shortest-delay offset generator uses a moving-window filter to select the samples of the phase offset signal having the shortest network propagation delay within the window. This shortest network propagation delay filter minimizes the effect of network jitter under the assumption that queuing delays account for most of the network jitter. The addition of this filtered phase offset signal to a free-running local clock creates a time reference that is synchronized to the remote clock at the source thus allowing for the transport of audio, video, and other time-sensitive real-time signals with minimal latency.

In an improved system for receiving digital voice signals from a data network, a jitter buffer manager monitors packet arrival times, determines a time varying transit delay variation parameter and adaptively controls jitter buffer size in response to the variation parameter. A speed control module responds to a control signal from the jitter buffer manager by modifying the rate of data consumption from the jitter buffer, to compensate for changes in buffer size, preferably in a manner which maintains audio output with acceptable, natural human speech characteristics. Preferably, the manager also calculates average packet delay and controls the speed control module to adaptively align the jitter buffer's center with the average packet delay time.

Presented are systems and methods that provide a unified end-to-end detection pipeline for object detection that achieves impressive performance in detecting very small and highly overlapped objects in face and car images. Various embodiments of the present disclosure provide for an accurate and efficient one-stage FCN-based object detector that may be optimized end-to-end during training. Certain embodiments train the object detector on a single scale using jitter-augmentation integrated landmark localization information through joint multi-task learning to improve the performance and accuracy of end-to-end object detection. Various embodiments apply hard negative mining techniques during training to bootstrap detection performance. The presented are systems and methods are highly suitable for situations where region proposal generation methods may fail, and they outperform many existing sliding window fashion FCN detection frameworks when detecting objects at small scales and under heavy occlusion conditions.

Systems and methods are provided for receiving and transmitting an MPEG2 transport stream (TS) in a real-time protocol (RTP)/user datagram protocol (UDP)/Internet protocol (IP) packet. The receiving method comprises: receiving an IP packet via an IP network, having a variable transmission delay; accessing a timestamp carried in a RTP packet; linking the timestamp with a program clock reference (PCR) MPEG2TS carried in the RTP packet payload; and, using the timestamp to eliminate variable transmission delay jitter, associated with the PCR MPEG2TS. In one aspect of the method, the timestamp has a resolution of greater than 500 nanoseconds (ns), so that the variable transmission delay jitter, associated with the PCR MPEG2TS can be reduced to a jitter of less than 500 ns.

Over-terminating the differential mode impedance of a differential transmission line, such as an INFINIBAND™ cable, at the receiving end, improves the differential signal integrity for typical variations in termination network impedance component (e.g., resistor) and transmission line characteristics. Eye opening of the differential signal can be made larger with reduced attenuation but increased jitter compared to under-terminating the differential mode impedance. Because the differential signal quality (larger eye opening) is improved, data can be transmitted over a longer transmission line with the same transmitter and receiver.

A system and method for streaming media on a wireless network is provided. In one aspect, the system includes a streaming device driver; a multidestination wireless streaming protocol module, a multitank level algorithm module, a synchronization mechanism module, and an autoequalizing speaker and drive system module. The system and method minimize latency, dropout, jitter, and synchronization issues associated with media streaming while optimizing performance and output quality of the media.

A Next Generation Network (NGN) resource management system and method includes a network topology discovery mechanism at the scale of an administrative domain. Information about nodes and links, such as bandwidth, delay, jitter, name and description of devices is collected and stored in a database by way of a protocol. The protocol is notifications-based, which involves each node device (e.g., a switch, router etc.) notifying its presence to its neighboring node.

A system for identifying chronic performance problems on data networks includes network monitoring devices that provide measurements of performance metrics such as latency, jitter, and throughput, and a processing system for analyzing measurement data to identify the onset of chronic performance problems. Network behavior is analyzed to develop a baseline performance level (e.g., computing a mean and variance of a baseline sample of performance metric measurements). An operating performance level is measured during operation of the network (e.g., computing a mean and variance of an operating sample of performance metric measurements). A chronic performance problem is identified if the operating performance level exceeds a performance threshold and a difference between the baseline performance level and the operating performance level is determined to be statistically significant. A t-test based on the baseline mean and variance and the operating mean and variance can be used to determine if this difference is significant.

A “communications rate controller” provides various techniques for maximizing a quality of real-time communications (RTC) (including audio and/or video broadcasts and conferencing) over multi-hop networks such as, for example, the Internet. Endpoints in such networks generally communicate via a segmented path that extends through one or more routers between each endpoint. Maximization of conferencing quality is generally accomplished by providing in-session bandwidth estimation across segments of the network path between endpoints (i.e., communication/conference participants) in combination with a robust non-oscillating dynamic rate control strategy for maximizing usage of available bandwidth between RTC endpoints. Further, the dynamic rate control techniques provided by the communications rate controller are designed to prevent degradation in end-to-end delay, jitter, and packet loss characteristics of the RTC.

Methods and apparatus for optimum matching of a traffic profile with an individual traffic flow using flow bifurcation and analysis. Bifurcation and duplication of packets that make up a flow received at an ingress element are forwarded to each of egress traffic and computation processor resources, such that egress traffic operations and traffic analysis operations may be performed concurrently without introducing jitter or delay in either bifurcated processing path. The traffic analysis includes maintaining flow statistics, flow stateful information and classifying the flow as a particular application traffic type. The optimum traffic profile for this application traffic type is then selected and applied to the individual flow. The traffic analysis data is forwarded to ingress and egress processing elements in real time, and ingress and egress traffic processing operations are dynamically adjusted in view of the traffic analysis data.

An adaptive buffering scheme allows more effective media transport and buffering. In one aspect of the adaptive buffering scheme, buffering parameters are adapted to different media characteristics, such as media play commands or the amount of encoding/transcoding required for the particular media stream. In another aspect of the adaptive buffering scheme, buffering is adapted to different transmission or memory conditions, such as transmission rate, packet jitter, or the amount of available buffer memory. In one example, the adaptive buffering is supported using Real Time Streaming Protocol (RTSP), and/or Real Time Transport Protocol (RTP) and associated Real Time Control Protocol (RTCP), and/or Session Description Protocol (SDP) messages.

A communications system can provide methods of dynamically interleaving streams, including methods for dynamically introducing greater amounts of interleaving as a stream is transmitted independently of any source block structure to spread out losses or errors in the channel over a much larger period of time within the original stream than if interleaving were not introduced, provide superior protection against packet loss or packet corruption when used with FEC coding, provide superior protection against network jitter, and allow content zapping time and the content transition time to be reduced to a minimum and minimal content transition times. Streams may be partitioned into sub-streams, delivering the sub-streams to receivers along different paths through a network and receiving concurrently different sub-streams at a receiver sent from potentially different servers. When used in conjunction with FEC encoding, the methods include delivering portions of an encoding of each source block from potentially different servers.

A method for stream synchronization of real-time multimedia transport over a packet network, in which the multimedia includes a first stream and a second stream which are buffered and played through buffers. The method includes: measuring average transport delays of the first and second streams; measuring transport delay jitters of the first and second streams; and calculating a delay difference between the first and second streams which corresponds to the average transport delays and the transport delay jitters of the first and second streams, and setting buffer durations. A device for stream synchronization of real-time multimedia transport over a packet network is also provided.

In a speech communications network, continuous play of audio packets is achieved using a jitter buffer in a receiver. Audio packets are stored in the jitter buffer before decoding the audio packets into an audible output. When the level of stored audio packets approaches the full capacity of the jitter buffer, the rate at which the audio packets are played out of the jitter buffer is increased signaling a compression operation in the decoder. When the level of stored audio packets approaches an empty level of the jitter buffer, the rate which the audio packets are played out of the jitter buffer is reduced signaling an expansion operation in the decoder. Audio packets are not modified when the level of stored audio packets is within a predetermined range. A speed controller is provided to instruct the decoder to decode the audio packets according to either a compressed, expanded or normal audio packet status.

A method of processing low resolution input frames containing undersampled views of an optically imaged scene to produce a higher quality, higher resolution output frame. This method operates by obtaining a sequence of low resolution input frames containing different undersampled views of an optically imaged scene. With regard to each new low resolution input frame, the method involves the further steps of measuring a displacement between a previous low resolution input frame and a new low resolution input frame to sub-pixel precision to produce a measured displacement; coarsely registering a high resolution working frame with the new low resolution input frame based on the measured displacement; finely registering the new low resolution input frame with the high resolution working frame by expanding the new low resolution input frame into a new high resolution input frame based on the measured displacement; and merging the new high resolution input frame into the high resolution working frame to produce an output frame.

A digital receiver, that may be used to receive VSB/QAM digital television signals, includes an adaptive fine carrier recovery circuit that compensates for deviations in the carrier frequency or phase. The carrier recovery circuit de-rotates a signal including phase errors. Estimations of phase errors are filtered using a filter whose gain and bandwidth are adjusted adaptively. This allows the carrier recovery circuit to track phase/frequency offset without introducing significant jitter. In one embodiment, the receiver includes a DFE, and the adaptive carrier recovery circuit mitigates instability that might be associated with the DFE.

Methods and systems are provided for efficient handover of a media session between heterogeneous IP networks. A mobile device with Internet access can operate a software program to communicate with a corresponding node. The corresponding node may access the Internet through either (i) a NAT router or (ii) a public IP address. The mobile device establishes a media session with a corresponding node via the transmission of a first media stream and receipt of a second media stream, and a media control channel can optionally be implemented. The mobile device can acquire Internet access through a second IP address, and packets routed between the second IP address and the Internet may traverse a NAT router. A software routine can determine that handover of the media session from the first IP address to the second IP address is preferred. The mobile device may begin transmitting a third media stream to the corresponding node before the first media stream stops. The corresponding node can transmit a fourth media stream to the second IP address before terminating the transmission of the second stream to the first IP address. Software operating at the mobile device may include a handover predictive jitter buffer.

The invention discloses an electronic equipment and display device screen brightness adjusting method thereof, aiming at reducing jitter of the electronic equipment display device screen brightness. The method includes: a first and a second environment illuminance values are obtained, comparison is carried out to obtain an environment illuminance variation, whether the environment illuminance variation is over the preset threshold is judged to produce a judgement result; when the judgement result expresses that the environment illuminance variation is over the threshold, a target brightness value used for adjusting the electronic equipment display device screen brightness is determined according to the second environment illuminance value. The technical scheme of the invention reduces jitter of electronic equipment screen brightness and can prolong service life of screen, save system resource and power consumption and improve user experience.

The invention discloses a travel track data-based stroke identification method. The stroke identification method comprises the following steps of: dividing track points through speed, and combining the track points with the speed lower than a certain speed threshold value into candidate remaining positions; and combining the candidate remaining positions by utilizing a distance threshold value and a time threshold value so as to determine a true remaining point. According to the stroke identification method, the problems of positioning drift and jitter of mobile phone positioning track data is solved, and the stroke identification precision is high; when 300 seconds is taken as the time threshold value and 1,100 meters is taken as the distance threshold value, the recall ratio is 87.66 percent, and the precision ratio is 81.56 percent; and meanwhile, GPS (Global Position System) positioning track data can be analyzed through adjusting the distance threshold value and the time threshold value.