Patent application number | Description | Published |
20080228476 | ENTROPY CODING BY ADAPTING CODING BETWEEN LEVEL AND RUN LENGTH/LEVEL MODES - An audio encoder performs adaptive entropy encoding of audio data. For example, an audio encoder switches between variable dimension vector Huffman coding of direct levels of quantized audio data and run-level coding of run lengths and levels of quantized audio data. The encoder can use, for example, context-based arithmetic coding for coding run lengths and levels. The encoder can determine when to switch between coding modes by counting consecutive coefficients having a predominant value (e.g., zero). An audio decoder performs corresponding adaptive entropy decoding. | 09-18-2008 |
20080262855 | ENTROPY CODING BY ADAPTING CODING BETWEEN LEVEL AND RUN LENGTH/LEVEL MODES - An audio encoder performs adaptive entropy encoding of audio data. For example, an audio encoder switches between variable dimension vector Huffman coding of direct levels of quantized audio data and run-level coding of run lengths and levels of quantized audio data. The encoder can use, for example, context-based arithmetic coding for coding run lengths and levels. The encoder can determine when to switch between coding modes by counting consecutive coefficients having a predominant value (e.g., zero). An audio decoder performs corresponding adaptive entropy decoding. | 10-23-2008 |
20080312758 | CODING OF SPARSE DIGITAL MEDIA SPECTRAL DATA - An audio encoder/decoder provides efficient compression of spectral transform coefficient data characterized by sparse spectral peaks. The audio encoder/decoder applies a temporal prediction of the frequency position of spectral peaks. The spectral peaks in the transform coefficients that are predicted from those in a preceding transform coding block are encoded as a shift in frequency position from the previous transform coding block and two non-zero coefficient levels. The prediction may avoid coding very large zero-level transform coefficient runs as compared to conventional run length coding. For spectral peaks not predicted from those in a preceding transform coding block, the spectral peaks are encoded as a value trio of a length of a run of zero-level spectral transform coefficients, and two non-zero coefficient levels. | 12-18-2008 |
20080312759 | FLEXIBLE FREQUENCY AND TIME PARTITIONING IN PERCEPTUAL TRANSFORM CODING OF AUDIO - An audio encoder/decoder performs band partitioning for vector quantization encoding of spectral holes and missing high frequencies that result from quantization when encoding at low bit rates. The encoder/decoder determines a band structure for spectral holes based on two threshold parameters: a minimum hole size threshold and a maximum band size threshold. Spectral holes wider than the minimum hole size threshold are partitioned evenly into bands not exceeding the maximum band size threshold in size. Such hole filling bands are configured up to a preset number of hole filling bands. The bands for missing high frequencies are then configured by dividing the high frequency region into bands having binary-increasing, linearly-increasing or arbitrarily-configured band sizes up to a maximum overall number of bands. | 12-18-2008 |
20080319739 | LOW COMPLEXITY DECODER FOR COMPLEX TRANSFORM CODING OF MULTI-CHANNEL SOUND - A multi-channel audio decoder provides a reduced complexity processing to reconstruct multi-channel audio from an encoded bitstream in which the multi-channel audio is represented as a coded subset of the channels along with a complex channel correlation matrix parameterization. The decoder translates the complex channel correlation matrix parameterization to a real transform that satisfies the magnitude of the complex channel correlation matrix. The multi-channel audio is derived from the coded subset of channels via channel extension processing using a real value effect signal and real number scaling. | 12-25-2008 |
20090006103 | BITSTREAM SYNTAX FOR MULTI-PROCESS AUDIO DECODING - 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. | 01-01-2009 |
20090083046 | EFFICIENT CODING OF DIGITAL MEDIA SPECTRAL DATA USING WIDE-SENSE PERCEPTUAL SIMILARITY - 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. | 03-26-2009 |
20090112606 | CHANNEL EXTENSION CODING FOR MULTI-CHANNEL SOURCE - A multi-channel audio decoder reconstructs multi-channel audio of more than two physical channels from a reduced set of coded channels based on correlation parameters that specify a full power cross-correlation matrix of the physical channels, or merely preserve a partial correlation matrix (such as power of the physical channels, and some subset of cross-correlations between the physical channels, or cross-correlations of the physical channels with coded or virtual channels). | 04-30-2009 |
20090125315 | TRANSCODER USING ENCODER GENERATED SIDE INFORMATION - 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. | 05-14-2009 |
20090157203 | CLIENT-SIDE AUDIO SIGNAL MIXING ON LOW COMPUTATIONAL POWER PLAYER USING BEAT METADATA - 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. | 06-18-2009 |
20090210222 | Multi-Channel Hole-Filling For Audio Compression - Multi-channel hole-filling for audio compression is disclosed. Channel dependency groups (CDGs) are explicitly extracted based on channel transform information. Holes are detected within each CDG for each bark, and a CDG hole is identified as requiring filling as a particular section of frequency bandwidth larger than a predetermined hole bandwidth threshold and with all zero-value coefficients in all channels after quantizing. Bark weights are adjusted by multiplying the original bark weights with one calculated scalar so as to remove each detected CDG hole. | 08-20-2009 |
20090248424 | LOSSLESS AND NEAR LOSSLESS SCALABLE AUDIO CODEC - 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. | 10-01-2009 |
20090279605 | ENCODING STREAMING MEDIA AS A HIGH BIT RATE LAYER, A LOW BIT RATE LAYER, AND ONE OR MORE INTERMEDIATE BIT RATE LAYERS - A method of encoding an input video stream comprising a video component and an audio component is disclosed. The input video stream is split into a plurality of segments, each comprising a plurality of frames. Each of the segments is encoded as a low bit rate layer, a high bit rate layer, and one or more intermediate bit rate layers. The bit rate of the low bit rate layer is selected such that a network streaming the segment will always be able to stream the segment encoded as the low bit rate layer. The bit rate of the high bit rate layer is selected such that the segment is able to be decoded and played back at or above a quality threshold. The bit rates of the intermediate bit rate layers are produced by applying a bit rate factor to another bit rate. | 11-12-2009 |
20090282162 | OPTIMIZED CLIENT SIDE RATE CONTROL AND INDEXED FILE LAYOUT FOR STREAMING MEDIA - 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. | 11-12-2009 |
20090299754 | FACTORIZATION OF OVERLAPPING TRANFORMS INTO TWO BLOCK TRANSFORMS - 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. | 12-03-2009 |
20100080290 | FINE-GRAINED CLIENT-SIDE CONTROL OF SCALABLE MEDIA DELIVERY - Techniques and tools for adjusting quality and bit rate of multiple chunks of media delivered over a network are described. For example, each of the multiple chunks is encoded as multiple layers (e.g., a base layer and multiple embedded residual layers) for fine-grained scalability at different rate/quality points. A server stores the encoded data for the layers of chunks as well as curve information that parameterizes rate-distortion curves for the chunks. The server sends the curve information to a client. For the multiple chunks, the client uses the curve information to determine rate-distortion preferences for the respective chunks, then sends feedback indicating the rate-distortion preferences to the server. For each of the multiple chunks, the server, based at least in part upon the feedback, selects one or more scalable layers of the chunk to deliver to the client. | 04-01-2010 |
20100153822 | Constructing Forward Error Correction Codes - Construction and use of forward error correction codes is provided. A systematic MDS FEC code is obtained having a property wherein any set of contiguous or non-contiguous r packets can be lost during a data transmission of k data packets and r encoded packets and the original k packets can be recovered unambiguously. The systematic MDS FEC code is transformed into a (k+r, k) systematic MDS FEC code that guarantees at least one of the encoded packets is a parity packet. The starting systematic MDS FEC code may be Cauchy-based, and the transformation code derived from the starting Cauchy-based MDS FEC code allows for very efficient initialization, encoding and decoding operations. | 06-17-2010 |
20100195488 | OPTIMIZED TRANSPORT PROTOCOL FOR DELAY-SENSITIVE DATA - Transmission delays are minimized when packets are transmitted from a source computer over a network to a destination computer. The source computer measures the network's available bandwidth, forms a sequence of output packets from a sequence of data packets, and transmits the output packets over the network to the destination computer, where the transmission rate is ramped up to the measured bandwidth. In conjunction with the transmission, the source computer monitors a transmission delay indicator which it computes using acknowledgement packets it receives from the destination computer. Whenever the indicator specifies that the transmission delay is increasing, the source computer reduces the transmission rate until the indicator specifies that the delay is unchanged. The source computer dynamically decides whether each output packet will be a forward error correction packet or a single data packet, where the decision is based on minimizing the expected transmission delays. | 08-05-2010 |
20110035225 | ENTROPY CODING USING ESCAPE CODES TO SWITCH BETWEEN PLURAL CODE TABLES - An audio encoder performs adaptive entropy encoding of audio data. For example, an audio encoder switches between variable dimension vector Huffman coding of direct levels of quantized audio data and run-level coding of run lengths and levels of quantized audio data. The encoder can use, for example, context-based arithmetic coding for coding run lengths and levels. The encoder can determine when to switch between coding modes by counting consecutive coefficients having a predominant value (e.g., zero). An audio decoder performs corresponding adaptive entropy decoding. | 02-10-2011 |
20110035226 | COMPLEX-TRANSFORM CHANNEL CODING WITH EXTENDED-BAND FREQUENCY CODING - An audio encoder receives multi-channel audio data comprising a group of plural source channels and performs channel extension coding, which comprises encoding a combined channel for the group and determining plural parameters for representing individual source channels of the group as modified versions of the encoded combined channel. The encoder also performs frequency extension coding. The frequency extension coding can comprise, for example, partitioning frequency bands in the multi-channel audio data into a baseband group and an extended band group, and coding audio coefficients in the extended band group based on audio coefficients in the baseband group. The encoder also can perform other kinds of transforms. An audio decoder performs corresponding decoding and/or additional processing tasks, such as a forward complex transform. | 02-10-2011 |
20110196684 | BITSTREAM SYNTAX FOR MULTI-PROCESS AUDIO DECODING - 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. | 08-11-2011 |
20110216648 | CONGESTION CONTROL FOR DELAY SENSITIVE APPLICATIONS - 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. | 09-08-2011 |
20110219287 | REMOTE PRESENTATION OVER LOSSY TRANSPORT WITH FORWARD ERROR CORRECTION - 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. | 09-08-2011 |
20120069899 | ENTROPY ENCODING AND DECODING USING DIRECT LEVEL AND RUN-LENGTH/LEVEL CONTEXT-ADAPTIVE ARITHMETIC CODING/DECODING MODES - 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. | 03-22-2012 |
20120128010 | MINIMIZING NETWORK LATENCY IN INTERACTIVE INTERNET APPLICATIONS - A method and system that enhances a user's performance while interacting with an interactive internet application such as a Massively Multiplayer Online (MMO) game is provided. The network latency experienced by users participating in the MMO game is minimized by dynamically determining an optimal transmission action for a message generated by the MMO game. In one embodiment, determining the optimal transmission action for a message includes dynamically determining the optimal number of redundant Forward Error Correction (FEC) packets to add to a message prior to transmitting a message to a receiving device. The optimal number of FEC packets is determined based on a wide range of varying network conditions. | 05-24-2012 |
20120155262 | KERNEL AWARENESS OF PHYSICAL ENVIRONMENT - 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. | 06-21-2012 |
20120323584 | BITSTREAM SYNTAX FOR MULTI-PROCESS AUDIO DECODING - 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. | 12-20-2012 |
20130054544 | Content Aware Chunking for Achieving an Improved Chunk Size Distribution - 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). | 02-28-2013 |
20130114421 | ADAPTIVE BANDWIDTH ESTIMATION - 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. | 05-09-2013 |
20130124697 | OPTIMIZED CLIENT SIDE RATE CONTROL AND INDEXED FILE LAYOUT FOR STREAMING MEDIA - 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. | 05-16-2013 |
20130128735 | UNIVERSAL RATE CONTROL MECHANISM WITH PARAMETER ADAPTATION FOR REAL-TIME COMMUNICATION APPLICATIONS - 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. The URCMPA can also dynamically adapt operating congestion levels relative to competing TCP flows to maintain fair use of network resources. | 05-23-2013 |
20130279338 | CONGESTION CONTROL FOR DELAY SENSITIVE APPLICATIONS - 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. | 10-24-2013 |
20140156287 | BITSTREAM SYNTAX FOR MULTI-PROCESS AUDIO DECODING - 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. | 06-05-2014 |
20140244604 | PREDICTING DATA COMPRESSIBILITY USING DATA ENTROPY ESTIMATION - 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. | 08-28-2014 |
20140379910 | CONTROLLING BANDWIDTH ACROSS MULTIPLE USERS FOR INTERACTIVE SERVICES - Embodiments are directed to controlling bandwidth usage using a token-based crediting and debiting scheme and to allowing connections to temporarily exceed bandwidth allocations using token credits. In one scenario, a bandwidth managing service receives a request to establish a connection with a network. The connection is associated with various subscribers that are part of a subscription. The bandwidth managing service assigns tokens to the connection, which are distributed from a pool of tokens that represents a total available bandwidth for the network. The bandwidth managing service receives a data transfer request from a logical user to transfer data over the network connection, where the data transfer request includes at least some of the assigned tokens. The bandwidth managing service also allocates to the connection a specified amount of bandwidth commensurate with the number of assigned tokens provided in the data transfer request. | 12-25-2014 |