55 results about "Available bit rate" patented technology

A method for delivering content via adaptive streaming technique over multiple communication paths and a device implementing the method are disclosed.The method for providing a content to be rendered at a client device comprises at least a first past being identified by the address of a first interface and the address of a first server interface, and a second path being identified by the address of a second interface and and the address of a second server interface, the content being available under at least two versions having an encoding quality corresponding to a supported bit rate constraint, each of the at least two versions being temporally split into chunks corresponding to an identical rendered duration of the content, a chunk being identified by a time index i and by one of the supported bit rate.According to the invention the method comprises the steps of:measuring a first available bit rate on the first path and a second available bit rate on the second path;determining a requested bit rate among the supported bit rate from the measured first available bit rate and from the measured second available bit rate;sending a first request via the first path for receiving a first part of the chunk identified by the time index i and by the requested bit rate and a second request via the second path for receiving a second part of said chunk, said of the chunk being complementary; and each of said parts of the chunk having a size being computed from the determined available bit rate;receiving the requested first part via the first path and the requested second part via the second path.

A system, method and computer program product are described, by which a user, through a control interface in a subscriber interface device, requests from network intelligence both transport network bandwidth and type of service line mode as part of call setup. An improved system by which requests to change access line mode and bandwidth are communicated over transmission facilities via D channels through local switch fabric to local or remote network intelligence is detailed. Connections over the transmission facilities, established between user B channels and B channels of transport bandwidth or between user B channels and the Stat Mux function, are described. An improved system is summarized, in which connections directly to transport bandwidth support point-to-point n B+D upstream and m B+D downstream service, and connections via stat mux allow connectionless traffic to share a portion of the transport bandwidth on an as needed basis. The ability to request and change access to available bit rate (ABR), variable bit rate (VBR) and constant bit rate (CBR) services, in a connection-oriented (CONS) or connectionless (CLNS) network, are described.

This invention provides rate shaping in per-flow output queued routing mechanisms for available bit rate (ABR) service in networks having segmented ABR control loops.

A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. A speech encoder employing various encoding schemes based upon parameters including an available transmission bit rate. In addition, the speech encoder is operable to identify and apply an optimal encoding scheme for a given speech signal. The speech encoder may be applied code-excited linear prediction when the available bit rate is above a predetermined upper threshold. Pitch preprocessing, including continuous warping, may be applied when it is below a predetermined lower threshold. The encoder considers varying characteristics of the speech signal including the long term prediction mode of a previous frame, and a spectral difference between the line spectral frequencies of a current and a previous frame, a predicted pitch lag, an open loop pitch lag, a closed loop pitch lag, a pitch gain, and a pitch correlation.

In a method of transmitting video data over a packet network, an input video stream is fed to a compressor that generates a compressed output stream. The compressor determines a target bit rate for the output stream based on desired picture quality. A traffic management unit a determines the available bit rate for the compressed data stream based on network condition. The output is adjusted if the target bit rate exceeds the granted bit rate.

A method controls operation of a network wherein at least one coded information stream is delivered to at least one user via at least one link exposed to variable operating conditions. The method includes the steps of: monitoring the operating conditions of the at least one link, and selectively transcoding the at least one coded information stream by selectively varying at least one transcoding parameter as a function of the operating conditions monitored. The monitoring operating conditions of the at least one link includes evaluating a set of cost functions related to an available bit rate and to complexity of said coded information stream and the selective variation of at least one transcoding parameter includes selecting among a plurality of transcoding levels associated to different values of quantization, resolution and frame rate.

A receiver-driven streaming method is provided. The method includes receiving an original coded video stream from a transmitter 102 at a receiver 104. An available bit rate is measured at the receiver 104. A request for a different coded video stream and a switching stream is sent from the receiver 104 to the transmitter 102 based on the available bit rate. The requested switching stream is received from the transmitter 102 at the receiver 104. The requested coded video stream is received from the transmitter 102 at the receiver 104. According to one embodiment, switching streams 132a-132f are stored separately from coded video streams 130a-130c, and each switching stream 132a-132f and coded video stream 130a-130c is stored in a separate track from each other. In this way, switching streams 132a-132f maybe generated offline. Also, the receiver 104 controls the streams provided by the transmitter 102 by requesting specific switching streams 132a-132f and coded video streams 130a-103c.

Fairness is provided among network sessions contending for available bit rate (ABR) service in an asynchronous transfer mode network by determining congestion information on a per port basis. ABR cells are monitored to determine which rate information per outgoing port based on ABR cells leaving a switch. Using the ABR per port rate, the available capacity, and the input rate to output queues, a link congestion parameter (LCP) is determined. Resource Management (RM) cells traveling through the switch are monitored to determine whether the new LCP is lower than the explicit rate (ER) field in RM cell. If the new LCP is lower, the new LCP is stuffed into the RM cell. When the sender receives the LCP value, the sender output rate is adjusted based on the LCP. Because LCP is determined based on per port traffic flow, fairness is provided among session competing for available ABR capacity.

A method for the control of flows particularly within an ATM switch transmission network with a distributed architecture. The method distributes a specified number n of shapers to each input line card, a specific shaper being dedicated to the VBRnrt (Variable Bit Rate non-real time) category flows as a function of the totalized mean bit rate, and n-1 other shapers being adjusted as a function of the available bit rate (AVCR) of the output ports.

A system and method for managing data traffic through a path that includes a transmission link having a transport rate that is subject to change with time. In one embodiment the data traffic is shaped to ATM Forum's available bit rate (ABR) category of service. ABR traffic has an integrated resource management (RM) cell that returns congestion related information to the source over a feedback path. With this congestion information the source end point is able to tailor its traffic send or transmission rate in order to accommodate or take advantage of any change in transport rate encountered by the traffic due to dynamically changing rate over the transmission link.

A method estimates bit rates for sets of coding parameters, based on characteristics of a channel, and an end-to-end distortion is estimated based on signal statistics of the video. A particular set of coding parameters that satisfies an available bit rate of the channel and that minimizes the end-to-end distortion of the video is selected. Motion compensated multiple descriptions of the video are encoded according to the selected set of coding parameters.

A network device is arranged to perform frame chunking directed towards enabling fast video content starts on a client device. When a request for video content is received, characteristics of a connection to the client device, and the client device are used to determine a threshold bitrate that provides a defined amount of video content to the client device within a configurable amount of first play time. When a bitrate for the video content that satisfies the threshold bitrate is currently unavailable, then the first chunks or bytes of the video content may be optimized to satisfy the threshold bitrate. The optimized first chunks are then provided to the client device followed by the remaining video content at an available bitrate.

In an ATM satellite communication system, congestion parameters for the downlink from the satellite are measured in the satellite and are broadcast to the downlink terminals or to the downlink and uplink terminals linked by virtual channels extending through the satellite. At a downlink terminal or uplink terminal, the parameters broadcast from the satellite are used to calculate an explicit rate for a virtual channel in the downlink. The explicit rate calculated at the terminal is used by the source for the virtual channel to determine the rate of transmission of Available Bit Rate cells on the virtual channel.

A method allocates data rates to layers to be transmitted in a multiple input, multiple output communications system. An input data stream is demultiplexed into multiple layers. For each layer, determine statistics representing a capacity of the layer based on past observations of transmitting the layer through a channel. For each layer, determine an optimum data rate based on the statistics. For each layer, determine if the optimum data rate is less than a minimum data rate of a set of available bit rates, and, if true, selecting, for a particular layer, the minimum data rate from the set of available data rates, and otherwise, if false, selecting, for the particular layer, a closest data rate from the set of available data rates that is less than the optimum data rate.

The invention discloses a resource seizing method for a long term evolution (LTE) system during service congestion, comprising the following steps: determining 'a seizing service', 'a seized service queue' and 'candidate services' which are distributed with the priority lower than that of the seizing service in the seized service queue; and selecting m candidate services in the seized service queue, wherein, the accumulated sum of available bit rate (ABR) of the m candidate services is larger than the ABR of the seizing service in the acceptation and refusing directions of the seizing service, while the accumulated sum of the ABR of m-1 candidate services is smaller or equal to the ABR of the seizing service; if the m candidate services exist, performing forced release processing on the m selected candidate services, accepting the seized service to finish the seizing process; and if the m candidate services do not exist, determining that the seizing is failed, and putting the seizing service into a failure acceptation list. By utilizing the method, the services with high access bit rate and low priority can be selectively released, thus ensuring to seize the high-priority services accessed by the user.

A method for gracefully extending the range and / or capacity of voice communication systems is disclosed. The method involves the persistent storage of voice media on a communication device. When the usable bit rate on the network is poor and below that necessary for conducting a live conversation, voice media is transmitted and received by the communication device at the available usable bit rate on the network. Although latency may be introduced, the persistent storage of both transmitted and received media of a conversation provides the ability to extend the useful range of wireless networks beyond what is required for live conversations. In addition, the capacity and robustness in not being affected by external interferences for both wired and wireless communications is improved.

A communication device for gracefully extending the range and / or capacity of voice communication systems. The communication device includes a persistent storage device configured persistently store the voice media either generated when using the communication device or received over the network at the communication device. When the usable bit rate on the network is poor and below that necessary for conducting a live conversation, the communication device transmits voice media from persistent storage and stores received voice media in persistent storage at the available usable bit rate on the network. Although latency may be introduced during the back and forth transmissions of a conversation, the persistent storage of both transmitted and received media of a conversation provides the ability to extend the useful range of wireless networks beyond what is required for live conversations. In addition, the capacity and robustness in not being affected by external interferences for both wired and wireless communications is improved.

Systems and methods to deliver streaming video over a hybrid network are provided herein. The system includes a plurality of queues, each associated with a type of video frames and a classifier to store a packet of a plurality of packets in a respective queue of the plurality of queues based on a field in the packet that indicates the type of video frame in the packet. The system also includes a scheduler configured to selectively transmit or drop the packet based on an available bitrate for transmission and the type of video frame associated with the queue.

A method includes inserting a virtual circuit connectivity verification packet into aggregated traffic of m packets transmitted over multiple pseudowires in a network path, replying to the transmitted m packets with a virtual circuit connectivity verification packet with a packet loss indication when at least one packet loss is detected in an m packet group, and adjusting a rate of transmitting the aggregated m packets responsive to the packet loss indication. In the preferred embodiment the multiple pseudowires are one of constant bit rate and variable bit rate and the adjusting of the rate includes rate adjustment of variable bit rate pseudowires.

A method and an apparatus for controlling media delivery in a multimedia transport network. The method includes transmitting composition information (CI) on media assets within a media package to be transmitted, to a user equipment (UE), packetizing the media assets of the media package to packets, transmitting some of the packets to the UE, receiving a feedback message including information on an available bitrate determined by the UE from the UE while the some of the packets are transmitted, selecting media data, which can be transmitted according to the available bitrate, from the media package by using asset delivery characteristics (ADCs) of the media assets within the media package, and packetizing the selected media data to packets and transmitting the packets to the UE.

A method for gracefully extending the range and / or capacity of voice communication systems is disclosed. The method involves the persistent storage of voice media on a communication device. When the usable bit rate on the network is poor and below that necessary for conducting a live conversation, voice media is transmitted and received by the communication device at the available usable bit rate on the network. Although latency may be introduced, the persistent storage of both transmitted and received media of a conversation provides the ability to extend the useful range of wireless networks beyond what is required for live conversations. In addition, the capacity and robustness in not being affected by external interferences for both wired and wireless communications is improved.

A method, apparatus and computer program product are provided for delivering television programming to a subscriber. In one aspect the method includes receiving one or more channels of program content, and encoding each channel a plurality of times, each time at a different rate producing multiple rate encoded signals for each channel. The method can further include transmitting the multiple rate encoded signals to a central office over a network, determining an available bandwidth of a downstream link to a subscriber and a current bit rate available for transmission to the subscriber. The method can further include selecting a best available bit rate from the multiple rate encoded signals for a requested channel and transmitting the selected rate encoded signal to the subscriber.

The invention discloses a variable-bit-rate encoder based on AMR (adaptive multi-rate)-NB (narrow band) voice signals. The variable-bit-rate encoder comprises a preprocessing unit, a voice frame quality judging unit, an encoding mode selecting unit, a bit rate determining unit and a code-excited linear predictive encoding unit, wherein the preprocessing unit is used for numeralization of voice signals to form voice frames; the voice frame quality judging unit is used for judging quality grades of the current voice frames to provide respective encoding modes and target bit rates for the voice frames; the encoding mode selecting unit is used for selecting the voice frame encoding modes according to the quality grades; the bit rate determining unit is used for determining the target bit rates of the voice frames according to the encoding modes; the code-excited linear predictive encoding unit is used for encoding the voice frames according to the target bit rates of the voice frames to form encoded voice frames. The invention further discloses a variable-bit-rate decoder used correspondingly to the encoder, and discloses a variable-bit-rate encoding method and a variable-bit-rate decoding method. The variable-bit-rate encoder is lower in bit rate than AMR, capable of achieving variable bit rates according to voice frame content and capable of selecting required encoding bit rate modes according to voice frame content importance judgment by setting channel voice quality.

The logic for responding to the maximum bitrate request includes comparing the requested maximum bitrate and the maximum bitrate limit to the supported maximum bitrate. If the maximum bit rate limit is at least lower than the lowest supported maximum bit rate, then no communication service associated with the requested maximum bit rate is provided. If the maximum bitrate limit is at least as high as the lowest supported maximum bitrate, then the offered bitrate is chosen from a set of supported maximum bitrates that is at least as high as the requested maximum bitrate and not higher than the maximum bitrate limit, as long as such a supported maximum bitrate exists. If such a supported maximum bitrate does not exist, the offered bitrate is chosen to be the highest available supported maximum bitrate not higher than the maximum bitrate limit.

The invention provides a method and a device for addressing an audio file with a variable bit rate (VBR). The method comprises the following steps of: generating an addressing table according to file header information and frame header information of the audio file, wherein play time points and absolute addresses corresponding to the play time points are recorded in the addressing table, and one play time point is recorded at intervals of n frames; and searching a play time point in the addressing table according to addressing target time, and taking the absolute address corresponding to the play time point as an addressing target position. By the method and the device, the function of addressing the audio file with the VBR can be accurately and quickly realized to the greatest extent.

A method for evaluating an available bitrate over a path among one of at least two paths linking a first endpoint and a second endpoint, the path being called an evaluation path, is disclosed. The first endpoint is configured for sending data packets to the second endpoint, the second endpoint is being configured for sending at each data packet reception an acknowledgment message to the first endpoint through one of the at least two path selected at each data packet reception. The method comprises, at the first endpoint, steps of:inserting a flag in a first data packet out of the data packets to be transmitted to the second endpoint through the evaluation path;sending said first data packet through the evaluation path;measuring and recording a sending time of the first data packet from said first endpoint;receiving an acknowledgment message sent by the second endpoint through the evaluation path for acknowledging reception of the first data packet at said second endpoint;measuring and recording an arrival time of said acknowledgment message at said first endpoint;assessing the available bitrate from an interval between the sending time and the arrival time and from the size of said first data packet.A method for sending at a second endpoint an acknowledgement message over a path among one of at least two paths linking a first endpoint and said second endpoint is further disclosed. Associated device for evaluating an available bitrate and device for sending at a second endpoint an acknowledgement message are disclosed.

Various embodiments of the present invention relate to systems, devices and method of video encoding that select a quantization parameter set based on a global edge strength value and an available bitrate for a corresponding compressed frame. Quantization parameters are selected using a mathematical correlation between the global edge strength value of the I-frame and an available bitrate / maxim target frame size for the corresponding compressed frame.

A video transrater, and a transrater system and method. The transrater, system and method may select an optimal bit rate from among a plurality of available bit rates in order to obtain a particular video quality and / or format.