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: 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.

Abstract: 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.

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: 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).

Abstract: 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.

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: An audio encoder encodes a combined channel (e.g., a sum channel) for a group of plural physical audio channels. The encoder determines plural parameters for representing individual physical channels of the group as modified versions of the encoded combined channel. The plural parameters comprise ratios of power in each individual channel to power in the combined channel (e.g., a ratio of the power of a right channel to the power of the combined channel, and a ratio of the power of the left channel to the power of the combined channel). The plural parameters can include a complex parameter. The combined channel and the plural parameters facilitate reconstruction at the audio decoder of source channels. An audio decoder performs a forward complex transform on the multi-channel audio data and reconstructs plural channels from the multi-channel audio data. The decoder can maintain second-order statistics for the source channels.

Abstract: 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.

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 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: 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.

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: 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 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: An audio encoder performs frequency extension coding that comprises determining one or more shape parameters using a displacement vector that corresponds to a displacement of an even number (e.g., an even number of sub-bands between a sub-band in a baseband frequency range and a sub-band in an extended-band frequency range). The shape parameters can be determined on a per-audio-block basis. Restricting a displacement to an even number (in frequency extension coding or in other signal modulation schemes) can improve the quality of reconstructed audio. An audio encoder also can perform frequency extension coding that comprises determining one or more scale parameters at one or more audio blocks, and determining one or more anchor points for interpolating the one or more scale parameters.

Abstract: 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.

Abstract: 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.

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: An audio encoder performs entropy encoding of audio data. For example, an audio encoder determines whether a first code table in a group of plural code tables contains a code representing a first vector. If it does, the code is used, and otherwise the escape code from the first code table is used and the first vector is encoded using, at least in part, a second code table. An audio decoder performs corresponding entropy decoding.