摘要:
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).
摘要:
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.
摘要:
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).
摘要:
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.
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:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
An audio encoder performs entropy encoding of audio data. For example, an audio encoder determines a Huffman code from a Huffman code table to use for encoding a vector of audio data symbols, where the determining is based on a sum of values of the audio data symbols. An audio decoder performs corresponding entropy decoding.