摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.