Abstract: The invention relates to a method for controlling a bit-rate of a session in a packet-switched communications system where multiple sessions are set-up via a shared resource. A current bit-rate of the session is compared to a bit-rate range valid for the session to determine a distance from the position of the current bit-rate to one of the end-points of the bit-rate range. The current bit-rate is then adapted differently depending on the distance to the selected end-point. By controlling the bit-rate of each session that is set-up via the shared resource for applications that provide delay-sensitive services to users in this way the method and corresponding system and sender-receiver arrangements have the effect that rate adaptations can be distributed among the applications so that users share responsibility for rate adaptations.

Abstract: Methods and apparatus for utilizing information (e.g., metadata) relating to content in a multimedia distribution network. In one embodiment, the network comprises a hybrid fiber coax (HFC) cable network, and the information comprises metadata relating to the bitrate profile of deterministic content such as stored video. Content sources, or the network operator themselves, generate the metadata which may then be used by the operator to adjust or optimize the operation of the network; e.g., more efficiently allocate the program to a multiplex. Network apparatus adapted to implement the metadata functionality and related business methods are also disclosed.

Abstract: A method and apparatus for transmitting an audio signal over a communication channel comprising encoding the audio signal with an encoder 204 using a first sampling rate, filtering the audio signal using a first cut off frequency, the first cut off frequency being chosen in dependence upon the first sampling rate, and transmitting the encoded and filtered audio signal over the communication channel. The presence of a condition in which the sampling rate of the encoder 204 is to be switched to a second sampling rate at a switching time is determined and if the condition has been determined to be present, the cut off frequency used in the filtering step is gradually changed from the first cut off frequency to a second cut off frequency, the second cut off frequency being chosen in dependence upon the second sampling rate, such that the audio bandwidth of the transmitted signal changes gradually when the sampling rate is switched to the second sampling rate.

Abstract: The present invention concerns a method of calculating a filtered filling level of the input buffer of a gateway generating a data stream from a received data stream resisting the jitter of the stream received. It applies more particularly to a gateway receiving an MPEG (Moving Picture Experts Group) transport stream received according to the IP protocol (Internet Protocol) and retransmitted over an ASI interface (Asynchronous Serial Interface). The invention proposes a method of calculating a filtered filling level of the input buffer resisting the jitter of the stream received. The received stream passing through an input buffer, the method is based on a regulation as a function of the maximum level of the buffer evaluated over a period of time, making it possible to protect against the effects of the jitter introduced into the received stream.

Abstract: Methods and devices are disclosed for generating a reorganized transport datastream from transport data packets of an original transport datastream displaces the first transport data packets contained in an original transport datastream into determined new positions within the reorganized transport datastream. The new positions are determined dependent upon a first data rate at which the first transport data packets of the reorganized transport datastream are stored in second buffers and a second data rate at which the first transport data packets of the reorganized transport datastream are read out from the second buffers.

Abstract: A receiver can extract digital data from wireless radio frequency signal. The digital data contains at least two compressed digital audio data each associated with a different audio program. One compressed digital audio data is delivered in a first frequency band and the other compressed digital audio data in a second frequency band. A user can select the audio program. The receiver also receives a command instructing the receiver to receive digital audio data from a communication channel in a frequency band.

Abstract: A packet schedule timestamp is provided. A schedule information packet includes a packet schedule timestamp that represents the relative timing of data packets in a particular bitstream of a multiplexed transport stream. The schedule information packet further provides information about the timing of packets within a bitstream or multiple bitstreams before actual packet arrivals. Look-ahead processing techniques are provided to improve the efficiency of compressed bitstream storage and delivery, such as program stream to transport stream conversion and statistical multiplexing. The packet schedule information can be inserted into the bitstream it describes for transmission or storage with it. Other features, such as timing reconstruction, are also provided. The packet schedule information can be used to reconstruct the timing of the transport packets to their original state.

Abstract: A transmission device (100) outputs, to a reception device (200), a stored amount of packets in a first transmission buffer (105) just before a certain packet has been written to the first transmission buffer. A first reception buffer amount read unit (206) in the reception device reads a stored amount of packets in a first reception buffer just before the certain packet has been read from the first reception buffer. A storage amount addition unit (207) adds the stored amount of packets in the first transmission buffer and the stored amount of packets in the first reception buffer, and a correction unit (208) adjusts the frequency of a variable frequency oscillator in a reception timestamp timer 209 so that the resulting added value is a constant value.

Abstract: A method and apparatus are provided for transmitting, over a wireless link, video that is coded in multiple layers such that higher priority for the use of the transmission resources may be given to the lower layers of the layered video stream. The level of priority is signified by the session identification parameters, i.e., by the source and destination ip addresses and port numbers. These parameters are assigned to the base layer and to the one or more additional layers, respectively, such that at least two of said layers are assigned different parameter sets. At least the base layer is then transmitted on the wireless link.

Abstract: The present invention relates to a data transport container for transferring a number n of different types of data in an Internet Protocol network. The number n is superior or equal to 2, and data is generated as Ethernet Jumbo packets, wherein the data transport container is an Internet Protocol packet. The invention also relates to a source device and to a destination device respectively for transferring and receiving such a data transport containers. At least, the invention relates to a method for transferring n different types of data being generated as Gigabit Ethernet RTP-UDP-IP jumbo packets. Both devices are to be used in a method as mentioned above.

Abstract: Presented herein are systems, methods, and apparatus for displaying pictures. In one embodiment, there is presented a decoder system for decoding video data. The decoder system comprises a port and a transport processor. The port receives packets carrying encoded video data from a plurality of video streams. The transport processor adds a header to encoded video data from at least one of the packets. The header identifies a particular one of the plurality of video streams, wherein the at least one packet is from the particular one of the plurality of video stream.

Abstract: A method of assigning a priority for controlling a bit rate of a bitstream having a plurality of quality layers is provided. The method includes composing first quality layers for a reference picture, composing second quality layers for a current picture that is encoded with reference to the reference picture, and assigning a priority each of the first and second quality layers, wherein a low priority is assigned to a quality layer having a small influence on a video quality reduction of the current picture when the quality layer is truncated.

Abstract: Described embodiments provide for a reassembly system for processing an asynchronous transfer mode (ATM) cell of data into an ATM adaptation layer (AAL) packet. A preprocessor module identifies a first conversation identification of one or more minipackets in the ATM cell, and reassembles the one or more minipackets having the first conversation identification into a portion of the AAL packet. A preprocessor determines if a trigger has occurred. In response to a trigger, the preprocessor sends a portion of the reassembled minipackets having the first conversation identification to a destination processor.

Abstract: When a request is sent by a switching unit, relating to the use of at least one subset of a codec mode configuration for the establishment of a transcoder-free operation connection by a radio network controller, it is checked whether the at least one requested subset is supported by the radio network controller. If the at least one subset of at least one codec mode configuration is supported, a transcoder-free operation connection to the switching unit and to the communication terminal is established by the radio network controller. Furthermore, a message relating to the at least one subset of the codec mode configuration to be used is signalled from the radio network controller to the communication terminal for the transmission of data.

Abstract: For monitoring a data transmitter such as a streaming server, which supplies transmitted data in response to a request of a receiver, an examiner examines a data traffic at the input of a receiver or at the output of the receiver, the examiner being configured to ascertain whether the data traffic falls below a minimum threshold. If the minimum threshold is fallen below, a connection activator initiates a new connection between the receiver and the data transmitter. This serves to enable consistent data-transmitter monitoring, even for non-reliable point-to-point connections via the internet.

Abstract: A computer motherboard includes a display controller, a digital visual interface integrated (DVI-I) connector, and a switching unit. The switching unit includes four first terminals, two second terminals, and a control terminal. Two of the first terminals are respectively connected to clock and data pins of a first display data channel (DDC) of the display controller, the other two first terminals are respectively connected to clock and data pins of a second DDC of the display controller, and the second terminals are respectively connected to DDC clock and DDC data pins of the DVI-I connector. The control terminal is connected to a hot plug detect pin of the DVI-I connector to detect a voltage and correspondingly control the second terminals to be respectively connected to the two first terminals connected to the clock and data pins of the first DDC or the second DDC.

Abstract: A method of media transmission. The method includes providing at least one variable bit rate (VBR) video stream and multiplexing the at least one VBR video stream together with auxiliary data related to the at least one VBR video stream, in a constant bit rate (CBR) stream. The stream is transmitted to a first receiving unit, which is adapted to discard the auxiliary data as padding and use the at least one VBR stream and to a second receiving unit, which is adapted to extract the auxiliary data and use it in further handling of the at least one VBR stream.

Abstract: In one embodiment, the methods and apparatuses detect a network bandwidth; detect a sequence of frames; determine a motion based on the sequence of frames; set a target bit rate for the sequence of frames based on the network bandwidth; and set a frame rate for the sequence of frames based on the motion of the sequence of frames, wherein the target bit rate and the frame rate are utilized to process the sequence of frames.

Abstract: Traffic from a number of different service type users is multiplexed into minicells for transmission over an ATM link. Service type information is transmitted in a service specific field and is stored. This stored information is used to derive length information to delineate the different length user minicells. The technique occupies less bandwidth than the conventional methods for transmitting cell length information.

Abstract: A system and method are provided for transporting a serial stream via a lower speed network using multiple parallel paths. At a transmitter, an optical or electromagnetic waveform is accepted representing a serial stream of digital information, and unbundled into n virtual information streams. Each virtual information stream is divided into a sequence of segments. Each segment is encapsulated, creating a sequence of packets by adding a start indicator to the beginning of each segment, and a terminate indicator to the end of each segment. Each packet is disinterleaved across m lanes and reinterleaved into n branches of framed data. Optical or electromagnetic waveforms representing the framed data are transmitted via n network branches. A receiver is also provided, which essentially reverses the above-described transmission method.

Abstract: The CRC for the CPS Header of an ATM AAL2 cell is generated by a CRC generator which uses the 8 bits of the CID field to generate partial 5 bits CRCs which are loaded in a first table. The 6 bits LI field and 5 bits UUI field are added to the partial 5 bits CRC to form 16 bits. The CRC generator uses the 2.sup.16 bits to generate a second CRC table. The CRC for a particular CPS header is generated by correlating bits in the CID field, LI field and UUI field with the two tables.

Abstract: An apparatus for reliably transferring media over a network from a sending node to a receiving node. The sending node includes a media ascertaining element configured to ascertain if media to be transmitted from the sending node to the receiving node is either time-sensitive or not time-sensitive. The sending node also includes a transmission element that is configured to (i) transmit the non time-sensitive media from the sending node to the receiving node by adjusting the rate of transmission of the non time-sensitive media at the sending node based on network conditions and (ii) transmit the time-sensitive media from the sending node to the receiving node. The receiving node includes a media loss ascertaining element configured to ascertain at the receiving node if a predetermined acceptable network transmission loss level is met.

Abstract: A packet schedule timestamp is provided. A schedule information packet includes a packet schedule timestamp that represents the relative timing of data packets in a particular bitstream of a multiplexed transport stream. The schedule information packet further provides information about the timing of packets within a bitstream or multiple bitstreams before actual packet arrivals. Look-ahead processing techniques are provided to improve the efficiency of compressed bitstream storage and delivery, such as program stream to transport stream conversion and statistical multiplexing. The packet schedule information can be inserted into the bitstream it describes for transmission or storage with it. Other features, such as timing reconstruction, are also provided. The packet schedule information can be used to reconstruct the timing of the transport packets to their original state.

Abstract: When a request is sent by a switching unit, relating to the use of at least one subset of a codec mode configuration for the establishment of a transcoder-free operation connection by a radio network controller, it is checked whether the at least one requested subset is supported by the radio network controller. If the at least one subset of at least one codec mode configuration is supported, a transcoder-free operation connection to the switching unit and to the communication terminal is established by the radio network controller. Furthermore, a message relating to the at least one subset of the codec mode configuration to be used is signalled from the radio network controller to the communication terminal for the transmission of data.

Abstract: A receiver can extract digital data from wireless radio frequency signal. The digital data contains at least two sets of compressed digital audio data each associated with a different set of audio program. One set of compressed digital audio data is delivered in a first frequency band and the other set in a second frequency band. One of the sets is stored in a memory system and the remaining set is not stored in the memory system. A user can select one of the sets of compressed digital audio data for decompression and then conversion to analog audio signal.

Abstract: A bit rate control method may include calculating a quality measure of a current frame and a mean of quality measures of two or more previous frames in order to calculate a deviation of the quality measure of the current frame from the mean of the quality measures of the two or more previous frames; comparing the deviation to a critical value measure; and controlling a quantization parameter in response to a result of the comparison. A bit rate control apparatus may include a quality measure calculator that calculates the quality measure of the current frame and the mean of the quality measures of the two or more previous frames in order to calculate the deviation; a comparator that compares the deviation to the critical value measure; and a quantization parameter setting unit that controls the quantization parameter in response to the result of the comparison.

Abstract: In a method for controlling the data flow on a radio link between a WLAN base station and a WLAN mobile station, the data rate on the radio link is reduced for in each case one time period (36, 37) at time intervals (35). One preferred field of application for the method is to restrict general or unauthorized transmission of a telephone call via the Internet.

Abstract: A video signal output control method in an information processing apparatus which being to be locked to an external unit by a lock mechanism, the video signal output control method includes detecting the state of the lock mechanism, and stopping the output of the video signal to the external unit, when the lock mechanism is in the unlocked state.

Abstract: In one example embodiment, an apparatus is provided and includes a network device that identifies a data type of a payload of a data packet to be transmitted. The network device adapts one or more of data transmission rates according to the data type that was identified. A control portion of the data packet is used to identify the data type. In more specific embodiments, the data transmission rates are controlled by one or more link layer configuration parameters, which are controlled according to the identified data type of the data packet. In still other embodiments, the network device identifies data packets with payloads relating to Intra-coded picture frames (I-frames), Predicted pictures frames (P-frames), and Bi-directional predictive pictures frames (B-frames).

Abstract: A receiver contains a demodulator system that can extract digital data from wireless radio frequency signal. The digital data contains three sets of compressed digital audio data. Two sets of compressed digital audio data are delivered in a first frequency band and the other set in a second frequency band. The set in the second frequency band and one of the sets in the first frequency band are stored in a memory system and the remaining set is not stored in the memory system. A user can select one of these three sets of compressed digital audio data for decompression and then converted to analog audio signal.

Abstract: An apparatus for improving the ability of a recipient to review time-sensitive media upon receipt when network conditions are poor. The apparatus includes a media ascertaining element configured to ascertain if media to be sent from a sending node to a receiving node is time-sensitive. A bandwidth-ascertaining element ascertains if there is sufficient bandwidth for transmitting a full bit rate representation of the media at a first bit rate and at a first packetization interval. If the bandwidth is sufficient, a transmitter transmits the full bit rate representation of the media. If the bandwidth is insufficient, then the transmitter generates and transmits a reduced bit rate representation of the media.

Abstract: A time stamp adding device includes: PCR_PID detecting means for detecting a PCR_PID included in a PMT by searching TS packets included in a TS from the head of the TS including TS packets having no time stamp, input from the outside and transmitted in a MPEG2-TS format and analyzing the resulting PMT; PCR detecting means for detecting values of PCRs included between the head of the TS and the PMT and position information of the PCRs and detecting values of all PCRs in the TS packets between the PMT and the tail of the TS and position information of the PCRs in the TS; time interval calculating means for calculating a time interval for adding a time stamp and a value of the time stamp based on the PCR values; and time stamp adding means for adding the time stamps to the TS packets having no time stamp based on the time interval.

Abstract: A medium access control (MAC) device includes a first set of transmit pins, a second set of transmit pins, and a first encoder. The first set receives (i) first transmit data signals and (ii) first transmit control signals from a host. The first transmit data signals are received on a first subset of the first set. The second set outputs (i) second transmit data signals and (ii) second transmit control signals to a physical layer (PHY) device. The second transmit data signals are output on a second subset of the second set. The second subset includes fewer transmit pins than the first subset. The first encoder encodes two of the first transmit control signals received on two of the first set of transmit pins to generate one of the second transmit control signals output on one of the second set of transmit pins.

Abstract: A method is disclosed to set the speed of a network. The method supplies a network interconnected with a system controller and a plurality of switch domains, where each of those plurality of switch domains comprises one or more information storage devices and a switch domain controller, and sets by each of the plurality of switch domains a signaling rate for that switch domain. The method queries in-band by the system controller each of the plurality of switch domains for that switch domain's signaling rate, and provides in-band by each of the plurality of switch domains the signaling rate for that switch domain. The method provides in-band by the system controller to each of the plurality of switch domains a first speed selection command specifying a first network speed, and resets by each of the plurality of switch domains the signaling rate for that switch domain to the first network speed.

Abstract: A retransmission apparatus using an IP multicasting method for a DMB service including a plurality (n) of radio frequency (RF) receivers, each for receiving a DMB RF signal of a corresponding channel, a plurality (n) of DMB decoders, each for outputting a transport stream (TS) by decoding the DMB RF signal received from a corresponding RF receiver, a plurality (n) of slicers, each for processing the TS input from a corresponding DMB decoder; a plurality (n) of network interfacers, each for IP packetizing the TS sliced by a corresponding slicer; a network switch for receiving the IP packets from the plurality (n) of network interfacers, transferring the received IP packets to a multicasting processing unit, and receiving multicasting data from the multicasting processing unit, and the multicasting processing unit for receiving the IP packets from the network switch, creating a single piece of multicasting data by assigning multicasting addresses to the received IP packets, and outputting the created multicasti

Abstract: A digital broadcast transmitting/receiving system and a method for processing data are disclosed. The method for processing data may enhance the receiving performance of the receiving system by performing additional coding and multiplexing processes on the traffic information data and transmitting the processed data. Thus, robustness is provided to the traffic information data, thereby enabling the data to respond strongly against the channel environment which is always under constant and vast change.

Abstract: Methods for transmitting first packets encapsulated in second packets in a transmission system in which part of the first packets contain a first timing reference for synchronization of a receiver clock and at least part of said second packets contain a second timing reference for reducing transmission jitter of the second packets at the receiver is described. This method provides, at the transmitter, collecting first packets, determining whether a collected first packet contains a first timing reference and triggering transmission of a second packet encapsulating collected first packets including the first packet containing the first timing reference in response to a positive determination.

Abstract: In one embodiment, a VoIP device operable to estimate an upstream data rate for a network device is provided. The VoIP device includes a transceiver operable to transmit VoIP packets to and receive VoIP packets from the network device; and a logic engine configured to initiate a series of simulated VoIP streams through the network device to a VoIP call destination, the logic engine being further configured to determine when at least one of the simulated VoIP streams in the series is unsuccessful, the logic engine being further configured to estimate the upstream data rate for the network device based upon a data rate for those simulated VoIP streams preceding the unsuccessful simulated VoIP stream.

Abstract: In generation of a system header included in an program stream of extended MPEG-2 Systems, when a basic stream ID is an extension ID and basic buffer size information represents 0 byte, extension buffer size information expressed by a total of 24 bits of an extension ID, a marker bit and an extension stream ID and a total of 24 bits including an extension stream ID, a marker bit and extension buffer bound scale information and required for reproduction of an elementary stream to be multiplexed using a pair of an extension ID and an extension stream ID is generated and recorded.

Abstract: A network device includes a media gateway to receive and process a voice over digital subscriber line (VoDSL) communication to generate voice data in a predetermined format; a terminating unit to receive another type of voice communication and output voice data in the predetermined format; and a control unit to receive the generated voice data from the media gateway and the outputted voice data from the terminating unit, where the network device exchanges at least one of the VoDSL communication to the other type of voice communication or the other type of voice communication to the VoDSL communication.

Abstract: A system and method are provided for transporting a serial stream via a lower speed network using multiple parallel paths. At a transmitter, an optical or electromagnetic waveform is accepted representing a serial stream of digital information, and unbundled into n virtual information streams. Each virtual information stream is divided into a sequence of segments. Each segment is encapsulated, creating a sequence of packets by adding a start indicator to the beginning of each segment, and a terminate indicator to the end of each segment. Each packet is disinterleaved across m lanes and reinterleaved into n branches of framed data. Optical or electromagnetic waveforms representing the framed data are transmitted via n network branches. A receiver is also provided, which essentially reverses the above-described transmission method.

Abstract: A packet relaying apparatus comprises a payload type discrimination unit for discriminating the type of payload in each received packet, a payload reconstruction unit for dividing the payload in predetermined block size units and for reconstructing the payload in the predetermined block size units, and a controller for overall controlling the apparatus.

Abstract: A method of determining the sequence of traversing nodes in a communication network is provided. The method comprises with each node, determining other nodes in the network it can reach in a single hop. Creating a local sequence connection order based on the determined other nodes the node can reach in a single communication hop. Storing historical data of local sequence connection order information and determining a new local sequence connection order based in part on the historical data and communication information.

Abstract: A receiver contains a demodulator system that can extract digital data from wireless radio frequency signal. The digital data contains two sets of compressed digital audio data corresponding to two audio programs. The receiver also contains two decompressors. If a user selects a first audio program, a first compressor is used to decompress a first set of compressed digital audio data. If a user selects a second audio program, a second decompressor is used to decompress a second set of compressed digital audio data. The wireless radio frequency signal can be transmitted using a transmitter.

Abstract: Systems and methods for transmitting video are provided. The system determines the type of encoded video frames. Each of the video frames is then placed in a network transport packet. A priority level of the network transport packet is set based on the determined type of the video frame. The system can apply error correction coding to video frames of a lower priority level.

Abstract: An improved system and method for transmitting a multiplexed service stream over an error prone DVB-H channel. Media IP packets are priority labeled. For each time slice burst, the IP packets are grouped based upon the priority labels. MPE-FEC matrices are made for the different priority labels in each time slice burst. RSDT columns are computed such that the average service bit rate does not overshoot the maximum allowed service bit rate, and the protection increases with priority. The ADT and RSDT of the MPE-FEC matrices are then encapsulated into MPE-FEC sections.

Abstract: Methods and systems provide approaches to start code emulation prevention at a granularity higher than the bit level. By operating at a level other than the bit level, processing capability requirements on both the encoder and decoder side can be reduced. In accordance with one or more embodiments, a start code emulation prevention method looks for data patterns relative to fixed-size data portions larger than single bits. When a particular pattern is found, start code emulation prevention data is inserted to prevent start code emulation. The inserted data is larger than a single bit and, in some embodiments, comprises a byte. When a decoder decodes data that has had start code emulation prevention data inserted, it can easily identify legitimate start codes and then can remove the start code emulation prevention data to provide the original data that was protected.

Abstract: A transcoder is provided for converting copyright information. The transcoder includes a separator configured to extract a video transport stream packet including a first video stream having copyright information, a non-video transport stream packet, and timing information from a first transport stream. A memory stores the timing information. The conversion unit generates a second video stream from the first video stream by converting the copyright information. A multiplexer multiplexes a video transport stream packet including the second video stream and the non-video transport stream packet based on the stored timing information to generate a second transport stream.

Abstract: A receiver contains a demodulator system that can extract digital data from wireless radio frequency signal. The digital data contains at least three sets of compressed digital audio data each corresponding to an audio program. At least two of the three sets of compressed digital audio data are stored in storage. Each of the stored sets of compressed digital data is associated with a set of index data. The receiver contains an output device allowing a user to perceive the sets of index data. The receiver also contains a digital to analog converter. If a user selects one of the sets of compressed digital audio data, the selected compressed digital audio data is decompressed and converted to analog audio data.

Abstract: A procedure is provided for transmitting packets of variable length, such as AAL2 packets and permitting the transmission of cells such as ATM cells, in time-division circuits such as E1/T1 frames, each divided into a number of time slots, each occupied by an eight-bit byte. The consecutive eight-bit bytes of each packet of variable length and of each cell are located in the slots of consecutive time of at least one group of time slots of the frames. The first eight-bit byte of a packet or a cell is placed at a time slot marked by a pointer occupying a specific time slot of the corresponding group of time slots.