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:
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.
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:
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:
The present invention relates to motion estimation and compensation. For example, a screen capture encoder performs motion estimation that is adapted to screen capture video in various respects. For example, the motion estimation uses a distortion measure based upon the count of equal/unequal pixels in two regions, sub-samples the distortion measure to speed up motion estimation, and/or uses a search pattern that prioritizes types of motion common in screen capture video. Or, a screen capture decoder performs motion compensation that is adapted to screen capture video in various respects. For example, the decoder performs the motion compensation for pixels with different values at corresponding locations in a current frame and a reference frame, but not for all pixels of the current frame. Alternatively, an encoder/decoder performs the motion estimation/compensation to compress/decompress other kinds of content.
摘要:
For encoding of mixed-content images containing palettized and continuous-tone content, continuous tone content regions in the image are detected and separated. Continuous tone content segmentation classifies pixels as continuous tone content by counting a number of unique pixel values within a pixel neighborhood. Pixels whose count exceeds a threshold are classified as continuous tone content. The technique further scans the image for regions of high continuous tone pixel density. The segmented continuous-tone and palettized content can be encoded separately for efficient compression, and then reassembled at decompression.
摘要:
The present invention relates to regulating the quality and/or bitrate of content within mixed content video when the video is compressed subject to a bitrate constraint. For example, a screen capture encoder encodes palletized content within a frame of screen capture video. Subject to an overall bitrate constraint, the encoder then allocates bits for continuous tone content within the frame. By controlling the allocation of bits used to encode the continuous tone content, the encoder regulates bitrate for the continuous tone content. This in turn can allow the encoder to regulate spatial quality and/or overall temporal quality for the video. In one scenario, for screen capture video encoded to a (relatively) constant overall bitrate, the screen capture encoder reduces the bitrate (and quality) of the continuous tone content, instead spending bits to increase the overall frame rate of the video.
摘要:
A system and method for correcting errors and losses occurring during a receiver-driven layered multicast (RLM) of real-time media over a heterogeneous packet network such as the Internet. This is accomplished by augmenting RLM with one or more layers of error correction information. This allows each receiver to separately optimize the quality of received audio and video information by subscribing to at least one error correction layer. Ideally, each source layer in a RLM would have one or more multicasted error correction data streams (i.e., layers) associated therewith. Each of the error correction layers would contain information that can be used to replace lost packets from the associated source layer. More than one error correction layer is proposed as some of the error correction packets contained in the data stream needed to replace the packets lost in the associated source stream may themselves be lost in transmission. A preferred process for generating the error correction streams involves the use of a unique adaptation of the Forward Error Correction (FEC) techniques. This process encodes the transmission data using a linear transform which adds redundant elements. The redundancy permits losses to be corrected because any of the original data elements can be derived from any of the encoded elements. Thus, as long as enough of the encoded data elements are received so as to equal the number of the original data elements, it is possible to derive all the original elements.
摘要:
Methods and apparatus for processing video data that is divided into frames are presented. In one aspect, this relates to a method for processing video data that is divided into frames. The video data includes a current frame, which has an associated current macroblock, and an adjacent frame, which also has an associated macroblock. The method for processing video data involves obtaining an uncompressed current block that is a part of the current macroblock and an adjacent block that is part of the adjacent macroblock, and calculating a distance between the uncompressed current block and the adjacent block. It is determined whether the distance between the uncompressed current block and the adjacent block is acceptable. If the distance is unacceptable, then the motion between the uncompressed current block and the adjacent block is estimated, and the uncompressed current block is adaptively compressed.