5388 results about "Bit rate" patented technology

Coding techniques for a (e.g., OFDM) communication system capable of transmitting data on a number of "transmission channels" at different information bit rates based on the channels' achieved SNR. A base code is used in combination with common or variable puncturing to achieve different coding rates required by the transmission channels. The data (i.e., information bits) for a data transmission is encoded with the base code, and the coded bits for each channel (or group of channels with the similar transmission capabilities) are punctured to achieve the required coding rate. The coded bits may be interleaved (e.g., to combat fading and remove correlation between coded bits in each modulation symbol) prior to puncturing. The unpunctured coded bits are grouped into non-binary symbols and mapped to modulation symbols (e.g., using Gray mapping). The modulation symbol may be "pre-conditioned" and prior to transmission.

Metadata for splicing of an encoded digital motion video stream (such as an MPEG Transport Stream) is prepared in real time while recording at the encoding bit rate and faster than encoded bit rate for off line encoding independent of the bit rate and mechanisms for ingestion of the data stream into data storage. Preprocessing is performed during a metered file transfer protocol (FTP) and includes pseudo real-time encoding. The preprocessing includes Group of Pictures (GOP) level pre-processing of splicing In Points and results in an intimate linkage between metadata and the file system in which the video data is stored. The preferred file system enables access to metadata in parallel to writing the data on disk. The pre-processing is performed simultaneous to writing the data to the disk using a carousel type buffer mechanism.

A method of billing a variable bit rate communication between a first terminal and a distant terminal to a broadband subscriber permits changing billing parameters during a call in real time in response to user inputs including user requested changes in quality of service, changes in data rate and changes in preferred service provider. A variable bit rate communication to be billed has a variable quality of service related to the degree of utilization of a plurality of different networks. The billing method comprises the steps of i.) receiving user identification data at a first terminal and data representing a required bit rate and a default quality of service selected by the user, ii.) verifying the user identification data to be associated with the broadband service subscriber, iii.) determining least cost alternative network resources available for achieving the communication at the user selected default quality of service and the required bit rate, iv.) determining cost data associated with the network resources, v.) outputting to the user a least cost for the communication according to their selected default quality of service and alternative least cost network resources, vi.) coupling the first terminal and the distant terminal via the least cost determined network resources at the default quality of service and the required bit rate responsive to user authorization and vii.) billing for the communication at the default quality of service and according to the required bit rate after the termination of the communication.

Streaming video data from the video title is delivered to a client from a central server and a proxy server. To reduce the backbone bandwidth required, a portion of the video title is cached at the proxy server where the portion cached is distributed over the video title so as to reduce the peak bit rate required for the central server to send the remaining portion of the video title. The amount cached at the proxy server may be adjusted in accordance with access profile of video titles to improve caching efficiency. The portion cached may be the basic layer in a scalable multilayer approach where only the basic layer may be sent to a client who has limited bandwidth while clients with higher bandwidth capability may also receive the enhancement layer(s) for improved video reception.

A moving picture variable bit rate coding apparatus receives digitized moving pictures and subjects them to coding according to a variable bit rate method using real time processing in response to the input of the pictures. The apparatus performs variable bit rate coding in real time by sequentially performing blocking, conversion processing, quantization processing, and generation of bit streams in response to the input of digitized moving picture data, and setting a quantization scale used for quantization corresponding to a quantity of generated bit streams to perform coding processing and control of quantization in parallel.

An improved data networking technique and apparatus using a novel physically distributed but logically shared and data-sliced synchronized shared memory switching datapath architecture integrated with a novel distributed data control path architecture to provide ideal output-buffered switching of data in networking systems, such as routers and switches, to support the increasing port densities and line rates with maximized network utilization and with per flow bit-rate latency and jitter guarantees, all while maintaining optimal throughput and quality of services under all data traffic scenarios, and with features of scalability in terms of number of data queues, ports and line rates, particularly for requirements ranging from network edge routers to the core of the network, thereby to eliminate both the need for the complication of centralized control for gathering system-wide information and for processing the same for egress traffic management functions and the need for a centralized scheduler, and eliminating also the need for buffering other than in the actual shared memory itself,—all with complete non-blocking data switching between ingress and egress ports, under all circumstances and scenarios.

If, in encoding and recording an AV stream, time_controlled_flag as the attribute information is set to 1, the relation between the time lapse of an AV stream and the amount of data bytes of the AV stream is linear within a preset error range. If the time_controlled_flag of the AV stream file is set to 1, and a certain time portion of the stream is erased by way of partial stream erasure, an unoccupied area corresponding to the erased time portion and which can be recorded at a bitrate indicated by TS_average_rate may be created on the disc.

A system and method have been provided for achieving simultaneous media playout in a network including a server and a plurality of clients. The method comprises: from a server, supplying a media stream to clients at a first bitrate (R1); determining the network delivery requirement; and, in response to the network delivery requirements, modifying the supply of the media stream. Determining the network delivery requirements includes determining either the buffering capacities of the clients, or the media streaming disruptions. To determine the buffering capacities of the dents, a first minimum client buffering capacity (C1) is determined by polling the clients for their respective buffering capacities, and selecting the first minimum buffering capacity (C1) to be equal to the client with the smallest buffering capacity.

Coding techniques for a (e.g., OFDM) communication system capable of transmitting data on a number of “transmission channels” at different information bit rates based on the channels' achieved SNR. A base code is used in combination with common or variable puncturing to achieve different coding rates required by the transmission channels. The data (i.e., information bits) for a data transmission is encoded with the base code, and the coded bits for each channel (or group of channels with the similar transmission capabilities) are punctured to achieve the required coding rate. The coded bits may be interleaved (e.g., to combat fading and remove correlation between coded bits in each modulation symbol) prior to puncturing. The unpunctured coded bits are grouped into non-binary symbols and mapped to modulation symbols (e.g., using Gray mapping). The modulation symbol may be “pre-conditioned” and prior to transmission.

A facility transport system for transporting high speed Ethernet data over digital subscriber lines. The system, referred to as 100BaseS, is capable of transmitting 100 Mbps Ethernet over existing copper infrastructure up to distances of approximately 400 meters. The system achieves bit rates from 25 to 100 Mbps in increments of 25 Mbps with each 25 Mbps increment utilizing a separate copper wire pair. Each pair used provides a bidirectional 25 Mbps link with four copper wire pair connections providing 4×25 Mbps downstream channels and 4×25 Mbps upstream channels. The system utilizes framing circuitry to adapt the 100BaseT input data signal to up to four separate output signals. A DSL Ethernet Port card couples the modem to each twisted pair used. Each DSL Ethernet Port card comprises modem transmitter and receiver circuitry for sending and receiving 100BaseS signals onto twisted pair wires. The system utilizes QAM in combination with frequency division multiplexing (FDM) to separate downstream channels from upstream channels and to separate both the downstream and the upstream channels from POTS and ISDN signals.

A wireless communication method and protocol, and wireless devices and systems for stimulation, are provided for communication between a wireless device and a charging device. During active wireless charging, communications (data transmission) from the wireless device to the charging device occurs via pulse loading the receive antenna of the receiving device. Because switching regulation in the receiving device may interfere with the communications, the switching regulation is disabled during a communications window. To further reduce the likelihood of misinterpretation of signals detected in the charging device resulting from the switching regulation or noise, the data bit rate of the pulse loading communications is maintained higher than the switching regulation frequency.

A system and method for processing of audio and speech signals is disclosed, which provide compatibility over a range of communication devices operating at different sampling frequencies and / or bit rates. The analyzer of the system divides the input signal in different portions, at least one of which carries information sufficient to provide intelligible reconstruction of the input signal. The analyzer also encodes separate information about other portions of the signal in an embedded manner, so that a smooth transition can be achieved from low bit-rate to high bit-rate applications. Accordingly, communication devices operating at different sampling rates and / or bit-rates can extract corresponding information from the output bit stream of the analyzer. In the present invention embedded information generally relates to separate parameters of the input signal, or to additional resolution in the transmission of original signal parameters. Non-linear techniques for enhancing the overall performance of the system are also disclosed. Also disclosed is a novel method of improving the quantization of signal parameters. In a specific embodiment the input signal is processed in two or more modes dependent on the state of the signal in a frame. When the signal is determined to be in a transition state, the encoder provides phase information about N sinusoids, which the decoder end uses to improve the quality of the output signal at low bit rates.

A method of billing a variable bit rate communication between a first terminal and a distant terminal to a broadband subscriber permits changing billing parameters during a call in real time in response to user inputs including user requested changes in quality of service, changes in data rate and changes in preferred service provider. A variable bit rate communication to be billed has a variable quality of service related to the degree of utilization of a plurality of different networks. The billing method comprises the steps of i.) receiving user identification data at a first terminal and data representing a required bit rate and a default quality of service selected by the user, ii.) verifying the user identification data to be associated with the broadband service subscriber, iii.) determining least cost alternative network resources available for achieving the communication at the user selected default quality of service and the required bit rate, iv.) determining cost data associated with the network resources, v.) outputting to the user a least cost for the communication according to their selected default quality of service and alternative least cost network resources, vi.) coupling the first terminal and the distant terminal via the least cost determined network resources at the default quality of service and the required bit rate responsive to user authorization and vii.) billing for the communication at the default quality of service and according to the required bit rate after the termination of the communication.

A system and method for caching files is disclosed. Popularity values are calculated for a plurality of files over a period of time. The popularity values are then used to determine which files should be cached at various remote sites. Once the caches are filled, the popularity values associated with the cached files are periodically compared with the popularity values of uncached content. Generally, if the popularity of an uncached file is greater than the popularity of a cached file, then the cached file will be replaced. However, numerous different variables may be factored into the caching determination including, for example, the size of the file, the required bitrate of the file, the identity of the owner of the file, and / or the type of file.

A method and apparatus for generating and transmitting bit rate conversion information to be utilized during bit rate conversion decisions and a method and apparatus for performing bit rate adaptation of media signals to use an available bandwidth of a channel, the media signals being associated with bit rate conversion information, the method for utilizing the bit rate conversion information including the steps of: receiving the media signals and the bit rate conversion information; the bit rate conversion information being indicative of a change in a bit rate of the media signals resulting from bit rate conversion; and converting the multiplexed processed media signals in response to the available bandwidth and the bit rate conversion information.

An apparatus, system, and method for maintaining a programming lineup of adaptive-bitrate content streaming is provided. The apparatus includes a timeline module configured to maintain a programming lineup of media content available over a network. The media content may comprise a plurality of streamlets. The apparatus also includes at least one data module configured to maintain multi-bitrate streamlet information. The system includes the apparatus and a client module configured to acquire content based upon the programming lineup provided by the timeline module. The method includes maintaining a programming lineup of media content available over a network, and maintaining multi-bitrate streamlet information.

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

An apparatus for multi-bitrate content streaming includes a receiving module configured to capture media content, a streamlet module configured to segment the media content and generate a plurality of streamlets, and an encoding module configured to generate a set of streamlets. The system includes the apparatus, wherein the set of streamlets comprises a plurality of streamlets having identical time indices and durations, and each streamlet of the set of streamlets having a unique bitrate, and wherein the encoding module comprises a master module configured to assign an encoding job to one of a plurality of host computing modules in response to an encoding job completion bid. A method includes receiving media content, segmenting the media content and generating a plurality of streamlets, and generating a set of streamlets.

A method for video playback switching in response to changing network conditions. The method includes accessing a server to retrieve respective index files for a low bit rate version and a high bit rate version of the video file, and instantiating a low bit rate media player and a high bit rate media. Playback of the video file is begun by the high bit rate media player streaming the high bit rate version from the server. Upon an indication of impeded network conditions, a transition point is selected, wherein the transition point indicates where downloading of the high bit rate version stops and where downloading of the low bit rate version begins. The low bit rate media player is then positioned to begin playback at the transition point. Playback of the video file is then switched to the low bit rate player upon encountering the transition point.

A novel method, system, and apparatus for efficient multi-block motion estimation in a digital signal compression and coding scheme. This invention selects only a few representative block sizes for motion estimation when certain favourable conditions occur, rather than using all available block sizes. This invention produces significantly reduced computational costs with virtually no sacrifice in visual quality and in bit-rate.

Providing a high-speed data service and voice service in a transmission system employing two binary, one quarternary (2B1Q) modulation / demodulation, using a high-speed data service remote terminal, a plurality of user data service and voice service terminals, and a multi-rate digital subscriber line (MDSL) terminal connected to the remote terminal through a twisted pair line, and to the user terminals. During downstream voice service, the remote terminal assembles and transmits an high bit rate digital subscriber line (HDSL) frame by including signaling signals for the voice service and signal processing mode information in a user-defined interval of the high bit rate digital subscriber line frame, to the multi-rate digital subscriber line terminal through the twisted pair line. During upstream voice service, the remote terminal receives the high bit rate digital subscriber line frame and transmits the signaling signals received to an exchange.

One embodiment of the invention sets forth an encoding server including components configured to encode a video stream associated with a content title for adaptive streaming. The video stream is first processed by a VC1 encoder to generate an encoded video stream comprising a multiple GOPs, each GOP including a key frame and having a different playback offset. The encoded video stream is then packaged such that the GOPs are stored in data packets of the packaged encoded stream. An SHI generator generates an SHI associated with the packaged encoded stream that includes a switch point associated with each GOP. Each switch point includes the playback offset associated with the corresponding GOP and the data packet storing the key frame of the corresponding GOP. The SHI associated with multiple packaged encoded video streams associated with the same content title and encoded to different playback bit rates have corresponding switch points.

A communication system selection algorithm (SSA) implemented by a mobile station chooses between available systems to select a system to serve the mobile station. During initialization, the SSA causes the mobile station to scan the environment and compare available communication systems to determine the best system to provide service. After an initial system is chosen, the SSA causes the mobile station to continuously, or at discrete time intervals, scan the environment for available systems, thus allowing for a seamless switch to an available system whenever a handoff is desired. The SSA chooses the best available system based on conditions such as bit rate and cost of each available system and applying preference rules defined by a service provider and / or user of the mobile station.

According to one embodiment, a circuit configured to form an output video stream includes a resolution modification circuit configured to receive a plurality of video frames from a frame buffer, and configured to modify resolution of the plurality of video frames, when the desired resolution for the output video stream is different than a resolution of the input video stream, the plurality of frames of data derived from an input video stream, a frame reducing circuit coupled to the resolution reducing circuit configured to reduce a number of video frames in the plurality of video frames from the resolution reducing circuit, when a desired frame rate for the output video stream is different than a frame rate of the input video stream, a depth reduction circuit coupled to the frame reducing circuit configured to reduce bit depth of the plurality of video frames from the frame reducing circuit, when a desired bit depth for the output video stream is different than a bit depth of the input video stream, and a rate reduction circuit coupled to the depth reduction circuit, configured to scale the plurality of video frames from the depth reduction circuit, in response to a desired bit rate for the output video stream, and an encoder coupled to the rate reduction circuit, configured to encode the plurality of video frames from the rate reduction circuit into the output video stream is also contemplated.

To reduce bandwidth of non-scalable MPEG-2 coded video, certain non-zero AC DCT coefficients for the 8x8 blocks are removed from the MPEG-2 coded video. In one implementation, high-frequency AC DCT coefficients are removed at the end of the coefficient scan order. This method requires the least computation and is most desirable if the reduced-bandwidth video is to be spatially sub-sampled. In another implementation, the smallest-magnitude AC DCT coefficients are removed. This method may produce an undesirable increase in the frequency of occurrence of escape sequences in the (run, level) coding. This frequency can be reduced by retaining certain non-zero AC DCT coefficients that are not the largest magnitude coefficients, and by increasing a quantization scale to reduce the coefficient levels. The reduced-bandwidth video can be used for a variety of applications, such as browsing for search and play-list generation, bit stream scaling for splicing, and bit-rate adjustment for services with limited resources and for multiplexing of transport streams.

The present invention is directed to a call admission controller that is operable to: (a) determine at least one of (i) a bandwidth utilization level for a first path including a first link; (ii) an available bandwidth level for the first path; and (iii) one or more Quality of Service or QoS metrics for the first path; (b) compare the at least one of (i) a bandwidth utilization level; (ii) an available bandwidth level; and (iii) one or more Quality of Service or QoS metrics to one or more selected thresholds to determine whether a new live voice communication may be set up with a first selected codec; and (iii) when a new live voice communication may not be set up with the first selected codec, perform at least one of the following operations: (i) select a second different codec from among a plurality of possible codecs for the new live voice communication, wherein the second codec has a lower bit rate than the first codec; (ii) change an existing live voice communication from the first codec to the second codec; and (iii) redirect the new live voice communication from the first path to a second different path, wherein the second path does not include the first link.

Entropy coding and decoding techniques are described, which may be implemented separately or in combination. For example, a video encoder uses two-layer run level coding to reduce bitrate for frequency transform coefficients in a quick and efficient manner, and a video decoder uses corresponding two-layer run level decoding. This two-layer coding / decoding can be generalized to more than two layers of run level coding / decoding. The video encoder and decoder exploit common patterns in run level information to reduce code table size and create opportunities for early termination of decoding. Using zoned Huffman code tables helps limit overall table size while still providing a level of adaptivity in encoding and decoding. Using embedded Huffman code tables allows the encoder and decoder to reuse codes for 8×8, 8×4, 4×8, and 4×4 blocks.

A system and method that integrates a plurality of media service systems offering different multimedia services such as media storage, syndication, delivery, and billing services. The system and method also provides automated file transcoding. In embodiment, a method of the present invention includes receiving a plurality of physical media files, organizing the plurality of physical media files so that different bit-rates and formats of a single source material are organized into a media database entity, receiving user specified delivery settings for the distribution of the physical media file, generating a release database entity storing the delivery settings of the physical media file, generating an address indicating the storage location of the release, and transmitting the address to a remote computing device.

A codec detects congestion in a packet network and responds via a session control protocol to re-negotiate codec-type and / or parameters with the receiving codec to reduce bit rate for supporting a session. Once the connection and session are established, encoded packets start flowing between the two codecs. A control entity sends and receives network congestion control packets periodically in the session. The congestion control packets provide a "heartbeat" signal to the receiving codec. When the network is not congested, all "heartbeat" packets will be passed through the network As network congestion increases, routers within the network discard excess packets to prevent network failure. The codecs respond to the missing packets by slowing down the bit rate or proceeding to renegotiate a lower bit rate via the session control protocol. If there are no missing packets, the codecs detect if the session is operating at the highest bit rate, and if not, re-negotiate a higher bit rate.