Abstract: The subject disclosure is directed towards predicting compressibility of a data block, and using the predicted compressibility in determining whether a data block if compressed will be sufficiently compressible to justify compression. In one aspect, data of the data block is processed to obtain an entropy estimate of the data block, e.g., based upon distinct value estimation. The compressibility prediction may be used in conjunction with a chunking mechanism of a data deduplication system.

Abstract: A computer-implemented method and system for determining documents that are nearest to a query are provided herein. The method includes constructing a vantage point tree based on a number of document vectors. The method also includes searching the vantage point tree to determine a number of nearest neighbor document vectors to a query vector by removing a portion of the document vectors from the vantage point tree based on one or more vantage points for each of a number of nodes in the vantage point tree and a specified search radius centered about the query vector.

Abstract: An encoder performs context-adaptive arithmetic encoding of transform coefficient data. For example, an encoder switches between coding of direct levels of quantized transform coefficient data and run-level coding of run lengths and levels of quantized transform coefficient data. The encoder can determine when to switch between coding modes based on a pre-determined switch point or by counting consecutive coefficients having a predominant value (e.g., zero). A decoder performs corresponding context-adaptive arithmetic decoding.

Abstract: An audio decoder provides a combination of decoding components including components implementing base band decoding, spectral peak decoding, frequency extension decoding and channel extension decoding techniques. The audio decoder decodes a compressed bitstream structured by a bitstream syntax scheme to permit the various decoding components to extract the appropriate parameters for their respective decoding technique.

Abstract: In various embodiments, methods and systems are disclosed for integrating a remote presentation protocol with a datagram based transport. In one embodiment, an integrated protocol is configured to support lossless or reduced loss transport based on Retransmission (ARQ) combined with Forward Error Correction (FEC). The protocol involves encoding and decoding of data packets including feedback headers and FEC packets, continuous measurement of RTT, RTO and packet delay, dynamically evaluating loss probability to determine and adjust the ratio of FEC, congestion management based on dynamically detecting increase in packet delay, and fast data transmission rate ramp-up based on detecting a decrease in packet delay.

Abstract: An encoder performs context-adaptive arithmetic encoding of transform coefficient data. For example, an encoder switches between coding of direct levels of quantized transform coefficient data and run-level coding of run lengths and levels of quantized transform coefficient data. The encoder can determine when to switch between coding modes based on a pre-determined switch point or by counting consecutive coefficients having a predominant value (e.g., zero). A decoder performs corresponding context-adaptive arithmetic decoding.

Abstract: Traditional audio encoders may conserve coding bit-rate by encoding fewer than all spectral coefficients, which can produce a blurry low-pass sound in the reconstruction. An audio encoder using wide-sense perceptual similarity improves the quality by encoding a perceptually similar version of the omitted spectral coefficients, represented as a scaled version of already coded spectrum. The omitted spectral coefficients are divided into a number of sub-bands. The sub-bands are encoded as two parameters: a scale factor, which may represent the energy in the band; and a shape parameter, which may represent a shape of the band. The shape parameter may be in the form of a motion vector pointing to a portion of the already coded spectrum, an index to a spectral shape in a fixed code-book, or a random noise vector. The encoding thus efficiently represents a scaled version of a similarly shaped portion of spectrum to be copied at decoding.

Abstract: An audio decoder provides a combination of decoding components including components implementing base band decoding, spectral peak decoding, frequency extension decoding and channel extension decoding techniques. The audio decoder decodes a compressed bitstream structured by a bitstream syntax scheme to permit the various decoding components to extract the appropriate parameters for their respective decoding technique.

Abstract: Described are techniques to use adaptive learning to control bandwidth or rate of transmission of a computer on a network. Congestion observations such as packet delay and packet loss are used to compute a congestion signal. The congestion signal is correlated with information about actual congestion on the network, and the transmission rate is adjusted according to the degree of correlation. Transmission rate may not adjust when packet delay or packet loss is not strongly correlated with actual congestion. The congestion signal is adaptively learned. For instance, the relative effects of loss and delay on the congestion signal may change over time. Moreover, an operating congestion level may be minimized by adaptive adjustment.

Abstract: In various embodiments, methods and systems are disclosed for a hybrid rate plus window based congestion protocol that controls the rate of packet transmission into the network and provides low queuing delay, practically zero packet loss, fair allocation of network resources amongst multiple flows, and full link utilization. In one embodiment, a congestion window may be used to control the maximum number of outstanding bits, a transmission rate may be used to control the rate of packets entering the network (packet pacing), a queuing delay based rate update may be used to control queuing delay within tolerated bounds and minimize packet loss, and aggressive ramp-up/graceful back-off may be used to fully utilize the link capacity and additive-increase, multiplicative-decrease (AIMD) rate control may be used to provide fairness amongst multiple flows.

Abstract: In various embodiments, methods and systems are disclosed for a hybrid rate plus window based congestion protocol that controls the rate of packet transmission into the network and provides low queuing delay, practically zero packet loss, fair allocation of network resources amongst multiple flows, and full link utilization. In one embodiment, a congestion window may be used to control the maximum number of outstanding bits, a transmission rate may be used to control the rate of packets entering the network (packet pacing), a queuing delay based rate update may be used to control queuing delay within tolerated bounds and minimize packet loss, and aggressive ramp-up/graceful back-off may be used to fully utilize the link capacity and additive-increase, multiplicative-decrease (AIMD) rate control may be used to provide fairness amongst multiple flows.

Abstract: An audio encoder encodes side information into a compressed audio bitstream containing encoding parameters used by the encoder for one or more encoding techniques, such as a noise-mask-ratio curve used for rate control. A transcoder uses the encoder generated side information to transcode the audio from the original compressed bitstream having an initial bit-rate into a second bitstream having a new bit-rate. Because the side information is derived from the original audio, the transcoder is able to better maintain audio quality of the transcoding. The side information also allows the transcoder to re-encode from an intermediate decoding/encoding stage for faster and lower complexity transcoding.

Abstract: A “Universal Rate Control Mechanism with Parameter Adaptation” (URCMPA) improves real-time communication (RTC) sessions in terms of delay, loss, throughput, and PSNR. The URCMPA automatically learns network characteristics including bottleneck link capacity, inherent queuing delay, inherent packet loss rates, etc., during RTC sessions. The URCMPA uses this information to dynamically adapt rate control parameters in a utility maximization (UM) framework. The URCMPA operates reliable RTC sessions across a wide range and combination of networks near full throughput rates while maintaining low operating congestion levels (e.g., low queuing delay and low packet loss). Examples of networks applicable for use with the URCMPA include, but are not limited to, combinations of mobile broadband (e.g., 3G, 4G, etc.), WiMAX, Wi-Fi hotspots, etc., and physical networks based on cable, fiber, ADSL, etc.

Abstract: An audio encoder/decoder uses a combination of an overlap windowing transform and block transform that have reversible implementations to provide a reversible, integer-integer form of a lapped transform. The reversible lapped transform permits both lossy and lossless transform domain coding of an audio signal having variable subframe sizes.

Abstract: It can be determined whether relative one way delay for data packets in a data stream exceeds a delay threshold. If so, then a delay congestion signal indicating that the relative one way delay exceeds the delay threshold can be generated. The delay congestion signal can be used in calculating an adaptive bandwidth estimate for the data stream. A packet loss rate congestion signal may also be used in calculating the bandwidth estimate. It can be determined whether a data stream of data packets is in a contention state. If the data stream is in the contention state, then an adaptive bandwidth estimate can be calculated for the data stream using a first bandwidth estimation technique. If the data stream is not in the contention state, then the bandwidth estimate for the data stream can be calculated using a second bandwidth estimation technique.

Abstract: A low computational power digital audio player achieves beat continuous transitioning between digital audio pieces based on beat metadata, which can be generated via offline processing on a higher computational power computer or via background or idle processing on the digital audio player. The digital audio player produces playlists of beat matching compatible songs based on the metadata, or pick lists of songs that are beat matching compatible with a currently playing song. By facilitating selection of songs with beat matching compatible tempos based on metadata, the beat continuous transitions can be achieved without altering the beat tempo of digital audio pieces, or with simple resampling.

Abstract: A video encoding system encodes video streams for multiple bit rate video streaming using an approach that permits the encoded bit rate to vary subject to a peak bit rate and average bit rate constraints for higher quality streams, while a bottom bit rate stream is encoded to achieve a constant chunk rate. The video encoding system also dynamically decides an encoding resolution for segments of the multiple bit rate video streams that varies with video complexity so as to achieve a better visual experience for multiple bit rate streaming.

Abstract: The subject disclosure is directed towards partitioning a file into chunks that satisfy a chunk size restriction, such as maximum and minimum chunk sizes, using a sliding window. For file positions within the chunk size restriction, a signature representative of a window fingerprint is compared with a target pattern, with a chunk boundary candidate identified if matched. Other signatures and patterns are then checked to determine a highest ranking signature (corresponding to a lowest numbered Rule) to associate with that chunk boundary candidate, or set an actual boundary if the highest ranked signature is matched. If the maximum chunk size is reached without matching the highest ranked signature, the chunking mechanism regresses to set the boundary based on the candidate with the next highest ranked signature (if no candidates, the boundary is set at the maximum). Also described is setting chunk boundaries based upon pattern detection (e.g., runs of zeros).

Abstract: A scalable audio codec encodes an input audio signal as a base layer at a high compression ratio and one or more residual signals as an enhancement layer of a compressed bitstream, which permits a lossless or near lossless reconstruction of the input audio signal at decoding. The scalable audio codec uses perceptual transform coding to encode the base layer. The residual is calculated in a transform domain, which includes a frequency and possibly also multi-channel transform of the input audio. For lossless reconstruction, the frequency and multi-channel transforms are reversible.

Abstract: An indexed file layout, comprising index information, is defined for segmented streaming of multimedia content. The index information can comprise program description information and streaming segment index information. In addition, the layout can comprise files containing streaming segments of the program, where the streaming segments are each encoded at one or more bitrates independently of other streaming segments of the program. The layout supports client switching between different bitrates at segment boundaries. Optimized client-side rate control of streaming content can be provided by defining a plurality of states, selecting available paths based on constraint conditions, and selecting a best path through the states (e.g., based on a distortion measure). In one client-side rate control solution states correspond to a specific bitrate of a specific streaming segment, and in another client-side rate control solution states correspond to a measure of client buffer fullness.