51results about How to "Less delay" patented technology

In a flex-rigid wiring board in which a rigid substrate formed from a rigid base material and a flexible substrate formed from a flexible base material are stack-joined and electrically connected to each other, the flexible substrate including a conductive layer having interconnecting electrode pads provided on at least one surface thereof, and the rigid substrate including a conductive layer having interconnecting electrode pads provided on at least one surface thereof in a position opposite to the interconnecting electrode pads on the rigid substrate, are locally connected electrically to each other with an anisotropic conductive adhesive layer interposed between conductive layers of substrate portions each including at least the interconnecting electrode pads. In this flex-rigid wiring board, transmission of high-frequency signals can be prevented from being delayed, noises can be suppressed, and an excellent electrical connection and connection reliability be assured.

Methods and apparatuses relating to optimization of radio network resources between user equipments operating in a cell and nodes in a cellular radio communication network are provided. Cross-layer information is read from higher level control plane protocol layer packets being transported, before or during lower level control plane protocol layer procedures being performed between a user equipment and a node or between two nodes The lower level control plane protocol layer procedures are optimized by using the analyzed control plane information received.

A multi-attribute handover judging method for a heterogeneous vehicle communication network comprises the following steps: a handover management entity reads information on a candidate network in a field network report buffer entity, and the diameter of the coverage region of the candidate network is calculated; a mobility prediction information management entity acquires position information and speed information of a mobile user, and sends the position information and the speed information to the handover management entity; possible residence time and throughput capacity in each neighbor access network of the mobile user is estimated, when the throughput capacity during handover is higher than the throughput capacity in a normal state, the candidate network is added in an alternative list; an optimal target access network is determined to be switched into the network by the handover management entity via an analytic hierarchy process and according to service quality requirements; when the intensity of the current switch-in network signal is lowered to a threshold value, a handover execution instruction is sent to carry out handover. The multi-attribute handover judging method provided by the invention avoids unnecessary handover when a traveling vehicle passes through a region covered by heterogeneous access network, reduces handover delay and ensures the continuity of the communication.

Systems and methods for synchronizing the playback of OTT or other time sensitive content on multiple playback devices is disclosed. The systems and methods include receiving time information based on a network time source in the playback devices. The playback clock in each playback device is set based upon the time information. Stream initiation information derived using the time information from the network time source is received by each of the playback device from the media provider. The playback devices use the stream initiation information to adjust the presentation time stamps of the frames of the media content in the stream.

Analog iterative decoders are provided that are based on the so-called min-sum algorithm (also referred to as max-sum or max-product, Max-Log-MAP or BP-based decoding) and can be used to decode powerful coding schemes such as low-density parity-check (LDPC) codes and turbo codes. The circuits can be implemented by standard CMOS technology, which means lower fabrication cost and/or simpler design compared to previously reported analog iterative decoders that are based on BiCMOS or sub-threshold CMOS technology. Soft information is passed among variable nodes and parity-check nodes. A low-voltage high-swing Max WTA circuit is also provided. The circuit can be implemented by short channel MOSFET transistors and yet provide a reasonably high degree of accuracy. Applications include soft computing, and analog signal processing, in general. A Min WTA circuit can also be built based on this circuit by subtracting the input currents from a large reference current.

A system and method, for converting a voltage input from a low voltage source to a voltage output at a high voltage source using a domino logic circuit design. An embodiment provides a low to high voltage conversion system. The system includes: a pull-up transistor coupled to a high voltage source for charging a node, when a precharge signal is received; a low voltage source used for setting an input voltage; a pull-down network for discharging the node depending, at least in part, on the input voltage; and an output voltage determined from the node.

An information processing device includes: a clipping unit configured to subject a plurality of images to clipping processing for clipping a part of an image as a partial image to clip a partial image from each of the plurality of images; a size correcting unit configured to correct an image size regarding each of a plurality of the partial images clipped by the clipping unit; and a synthesizing unit configured to connect the partial images of which the image sizes have been corrected by the size correcting unit to generate a synthetic image.

The present disclosure is directed to controlling a sending of a single data stream from a sending endpoint to a receiving endpoint. Both of the endpoints each have multiple physical interfaces connecting each endpoint to one or more networks. A buffer size is selected for a buffer at the receiving endpoint, and an estimation is made by the receiving endpoint of a fill rate of the buffer. The estimated fill rate is then compared with a target fill rate. Feedback is then sent by the receiving endpoint over one or more of the physical interfaces instructing the sending endpoint to reduce or increase a rate at which data is being sent, if the estimated fill rate is more than the target fill rate or the estimated fill rate is less than the target fill rate, respectively. The splitting and sending of data is then adjusted according to the feedback.

The present disclosure is directed to streaming a video from a sending endpoint to a receiving endpoint. Both of the sending endpoint and the receiving endpoint have multiple communication channels connecting the sending endpoint and the receiving endpoint to one or more networks, respectively, and the streaming video includes a plurality of intra-frame coded frames and a plurality of inter-frame coded frames. The sending endpoint splits the video stream into the plurality of intra-frame coded frames and inter-frame coded frames. The sending endpoint then sends different ones of the plurality of intra-frame coded frames to the receiving endpoint over more than one of the multiple communication channels having a connection-oriented protocol. Also, the sending endpoint sends different ones of the plurality of inter-frame coded frames to the receiving endpoint over more than one of the multiple communication channels having a connectionless-oriented protocol. The sent intra-frame coded frames and inter-frame coded frames are then received and recombined at the receiving endpoint.

A method and an apparatus for negotiating the used communication mode in a telecommunications system, wherein a connection is arranged between at least two system elements. The system elements exchange messages and check whether a first in-band signalling protocol is supported. The communication mode is selected according to the first signalling protocol if the system elements support the first signalling protocol. If the first signalling protocol is not supported, support for a tandem free operation (TFO) protocol is checked and communication between the system elements is continued according to TFO procedures if the system elements support TFO. If the first signalling protocol and TFO are not supported, the communication mode between the system elements may be selected according to a second in-band signalling protocol.

A method for correcting a shift error in a fused multiply add operation. The method comprises adjusting a normalized floating-point number before performing a shift error correction to produce an adjusted normalized floating-point number, and correcting a shift error in the adjusted normalized floating-point number. The correcting the shift error comprises shifting a mantissa of the adjusted normalized floating-point number in one direction. A fused multiply add module comprising a normalizer module, a compensation logic, and a round. The normalizer module is operable to normalize a floating-point number to produce a normalized floating-point number. The floating-point number is normalized based upon an estimated quantity of leading zeros. The compensation logic is operable to manage a correction of a shift error in the normalized floating-point number. The rounder is operable to correct the shift error with a mantissa shift in only one direction.

Various embodiments implement distributed block coding (DBC). DBC can be used for, among other things, distributed forward error correction (DFEC) of source data in communication systems or parity backup for error correction of source data in storage systems where the source data may be corrupted by burst errors. A distributed block encoder (DBE) encodes sequential source data symbols with a plurality of sequential block encoders to produce interleaved parity codewords. The interleaved parity codewords enable decoding of error-corrected source data symbols with a distributed block decoder (DBD) that utilizes a plurality of sequential block decoders to produce the error-corrected source data symbols. A distributed register block encoder (DRBE) and a distributed register block decoder (DRBD) can each be implemented in a single block encoder and a single block decoder, respectively, by using a distributed register arrangement.

A radio communication device that manages a network defines a predetermined transmission frame cycle, allocates in the frame specific access slots as timing with which the individual radio communication devices in the network concerned receive a transmission, puts the information of the allocation in a beacon signal, and transmits the signal. Each radio communication device performs a receiving operation at access slots specific to the device itself defined by the beacon signal. When information transmission is made between arbitrary communication devices in the network, access slots for the device that is to receive the information are used.

An improved inserter input system and method for transversely cutting a web of printed material into separate sheets, the web including a plurality of separated side-by-side sheets. A set of sheets on the web is transported to the cutting device. One or more of the sheets in the set belongs to a new collation for which sheets have not previously been cut. The system determines whether sufficient collation parking spots exist to accommodate a new collation. If there are no available collation parking spots, and if all of the sheets in the set belong to the new collation, then transverse cutting is delayed until an open collation parking spot becomes available. If there are no available collation parking spots, and if a subset of sheets belong to a prior collation, then the web is partially cut to separate only the sheet, or sheets, that belong to the prior collation. The cutting of the other sheet(s) is delayed until the open collation parking spot becomes available. If there is an available collation parking spot, then the cutting device transversely cuts the entire set of side-by-side sheets.

A transmitting apparatus arranges data in matrix, calculates error correction codes in the data's column direction, arranges the calculated error correction codes in matrix having the same number of columns as the data, attaches number information corresponding to a row number to each row having data or error correction code, encapsulates, into each packet, each row having number information and data or error correction code, plus error detection code detecting one or more bit errors at least in the number information and the data or error correction code in the packet, and transmits the packet. Receiving apparatus discards received packets having detected errors using the error detection code in each packet, checks packet losses according to number information, and restores lost data using the error correction codes. An error correction method compensates for packet losses and bit errors in a packet, realizing efficient transmission with less delay by simple configuration.

A method for scheduling packets from a plurality of radio bearers by a scheduler of a first layer includes calculating a virtual scheduling time for the radio bearers by using a packet delay variation of the radio bearers, and transmitting the packet of the radio bearer having the greatest virtual scheduling time from among the plurality of radio bearers to a second layer.

D trigger is composed of driving and triggering two parts of cascaded circuits. The driving circuit includes transmission gate, timeclock controlled inverting circuit and an inverter. Triggering circuit is a difference structure controlled by the timeclock controlled inverting circuits. On/off of transmission gate is controlled by timeclock singnal; when signal is in high, transmission gate is off; and when signal is in low, transmission gate is on. In on state of the transmission gate, input signal in high level is sent to trigger; when next timeclock signal in high level comes, transmission gate is turned to off; and timeclock controlled inverting circuit is turned to on to hold electrical level, and meanwhile trigger is flipped. Advantages are: simple circuit structure, few number of transistor, small area, about 40úÑ less than power consumption of traditional trigger, and about 20úÑ less than delay time. Moreover, difference input in second stage enhances performance for antinoise.

The invention discloses an audio and video stream direct broadcast method based on an HLS protocol. The method comprises a download thread and a play thread, wherein the download thread is used for managing a local cache directory, and requesting an m3u8 file and a video slice file for a content distribution end; the play thread is used for analyzing the m3u8 file in the cache directory, calculating slices needed to be played according to an EXMS value and a media serial number that has been played in the m3u8 file, and orderly playing; and the download thread and the play thread are executed asynchronously. The method of the invention guarantees the low delay of live broadcast, and increases the fluency of the live broadcast under the condition that the network environment is bad; besides, a management mechanism of a slice naming rule and the cache directory ensures that the local storage space occupation is less and no junk files are generated.

The present disclosure is directed to streaming a video from a sending endpoint to a receiving endpoint. Both of the sending endpoint and the receiving endpoint have multiple communication channels connecting the sending endpoint and the receiving endpoint to one or more networks, respectively, and the streaming video includes a plurality of intra-frame coded frames and a plurality of inter-frame coded frames. The sending endpoint sends different ones of the plurality of intra-frame coded frames and different ones of high priority inter-frame coded frames to the receiving endpoint over more than one of the multiple communication channels having a connection-oriented protocol. In addition, the sending endpoint sends different ones of non-high priority inter-frame coded frames to the receiving endpoint over more than one of the multiple communication channels having a connectionless-oriented protocol.

The present invention relates to a sound device (SD1), connected to a computer (P1), for handling of asynchronously transferred digital audio packets (5) on a network (LAN1). The computer has an interface (3) connected to a telephony application (1), a driver (D3) and a bus (4). The sound device (SD1) is connected (9) via the bus (4) and includes a software frame buffer (B2), codecs (C2) and an A/D-D/A converter (AD2), which is connected to in/out devices (10, 11, 12). The sound packets (5) are transferred asynchronously through the computer (P1), are buffered in the sound device frame buffer (B2), decoded in the codec (C2) and D/A converted into an analog signal for the in/out devices. Speech to the in devices (11, 12) is processed in a corresponding manner. Having the buffer (B2) close to the codec (C2) enables processing of the sound packets, e.g. with respect to the varying time delay in the computer (P1), restoring lost packets and producing replacement frames. The sound device (SD1) relieves the computer (P1) of the heavy workload of processing the sound packets (5).

A nonlinear distortion compensating device splits its operation into two processes. One is a compensation & amplification process for providing a receipt signal with a distortion compensating calculation and amplification before outputting the signal with phase and amplitude compensated. The other is an environmentally adapting process for modulating the signal demodulated from the input signal. Then in this process, a distortion estimating updater estimates a distortion amount using the modulated signal and a feed back signal of parts of the output supplied from the distortion compensating calculator, and updates a distortion compensating coefficient stored in the distortion compensating calculator.