Microsoft Technology Licensing, LLC

Neiversan Networks Co. LLC

Other

Micorsoft Corpration

Microsoa Corporation

Microsoft Corporation

Zhigu Holdings Limited

Henrique Sarmento Malvar

Patrice Y Simard

Darko Kirovski

Erin L Renshaw

Yacov Yacobi

Jack Wilson Stokes, Iii

Ivan Jelev Tashev

Thomas Walter Getzinger

Dinei Afonso Ferreira Florencio

James Russell Rinker

Paul England

Ross Garrett Cutler

Mariusz Hieronim Jakubowski

Dinei A Florencio

Bradford W Gillespie

Sridhar Srinivasan

Karin Strauss

Yong Rui

Chengjie Tu

Brian Kevin Guenter

Luis Henrique Ceze

Joseph Cox Ballantyne

David W Steinkraus

Qing Guo

Eric J Horvitz

Yongjun Wu

Zhengyou Zhang

Gary Joseph Sullivan

Alejandro Acero

Zicheng Liu

John Carlton Platt

Michael F Cohen

Sanjeev Mehrotra

Michael Dean Stokes

Eric H Rudolph

Wei-Ge Chen

Daniel F Plastina

James G Droppo

Li-wei He

Michael Lewis Seltzer

Hubert Van Hoof

Christopher J C Burges

Roy A Leban

Jerome E Lengyel

Li-Wei He

Chris J C Burges

Daniel J Sisolak

Byung-Jun Yoon

Diego Ariel Bendersky

Xing Fan

John E Unverferth, Iii

Cindy Marie Grimm

Sridhar Srinvasan

Ivan Tashev

Systems and methods to determine when a media is a high-fidelity reproduction of an original media from a trusted entity are disclosed. In certain aspects, systems and method for generating a fragile watermark are disclosed. The fragile watermark may be inserted into digital media in a manner such that the watermark cannot be identified if the media content is significantly altered. Media content may be subsequently analyzed to determine the presence of a fragile watermark. When the fragile watermark is present, provenance of the media content can be verified and an indication of provenance is provided to the user.

Media provenance certification via fragile watermarking

A memory chip for dynamic approximate storage includes an array of memory cells associated with at least two regions. The chip further includes at least one threshold register for storing values for thresholds for memory cells corresponding to each of the at least two regions; and control logic to programmatically adjust the values for the thresholds for the memory cells. A method of controlling a storage device for dynamic approximate storage includes modifying at least one value stored in a threshold register and associated with at least one cell in a region of a memory comprising at least two regions to apply a biasing for the at least one cell, wherein the biasing adjusts ranges for values in a cell.

Dynamic approximate storage for custom applications

A method of encoding data on single level or variable multi-level cell storage includes receiving a block of encoded data from an approximation-aware application and at least an importance attribute associated with the block of encoded data; and assigning the block of encoded data to a memory address or a particular region of a memory having at least three precision levels, based at least according to the importance attribute. The importance attribute indicates a relative sensitivity of bits of the block to errors in an output quality from decoding the encoded data. An approximation-aware application can be an image encoding application having a modified entropy encoding step that enables identification and splitting of bits into groupings according to sensitivity to errors.

Data encoding on single-level and variable multi-level cell storage

When packing a video frame of a higher-resolution chroma sampling format such as YUV 4:4:4 into frames of a lower-resolution chroma sampling format such as YUV 4:2:0, a computing device performs wavelet decomposition (or other band separation filtering) on sample values of chroma components of the higher-resolution frame, producing sample values of multiple bands. The device assigns the sample values of the bands to parts of the lower-resolution frames. During corresponding unpacking operations, a computing device assigns parts of the frames of the lower-resolution chroma sampling format to sample values of multiple bands. The device performs wavelet reconstruction (or other inverse band separation filtering) on the sample values of the bands, producing sample values of chroma components of the frame of the higher-resolution chroma sampling format. Band separation filtering can help improve quality of reconstruction when distortion has been introduced during encoding of the chroma components packed into low-resolution frames.

Band separation filtering / inverse filtering for frame packing / unpacking higher resolution chroma sampling formats

An 'Overcomplete Audio Coder' provides various techniques for overcomplete encoding audio signals using an MCLT-based predictive coder. Specifically, the Overcomplete Audio Coder uses unrestricted polar quantization of MCLT magnitude and phase coefficients. Further, quantized magnitude and phase coefficients are predicted based on properties of the audio signal and corresponding MCLT coefficients to reduce the bit rate overhead in encoding the audio signal. This prediction allows the Overcomplete Audio Coder to provide improved continuity of the magnitude of spectral components across encoded signal blocks, thereby reducing warbling artifacts. Coding rates achieved using these prediction techniques are comparable to that of encoding an orthogonal representation of an audio signal, such as with modulated lapped transform (MLT)-based coders. Finally, the Overcomplete Audio Coder provides a true magnitude-phase frequency-domain representation of the audio signal, thus allowing precise auditory models to be applied for improving compression performance, without the need for additional Fourier transforms.

Efficient coding of overcomplete representations of audio using the modulated complex lapped transform (MCLT)

An improved method and block transform for image or video encoding and decoding, wherein transformation and inverse transformation matrixes are defined such that computational complexity is significantly reduced when encoding and decoding. For example, in the two-dimensional inverse transformation of de-quantized transform coefficients into output pixel information during decoding, only four additions plus one shift operation are needed, per co-efficient transformation, all in sixteen-bit arithmetic. Transformations provide correct results because quantization during encoding and de-quantization (sixteen bit) during decoding, via the use of one of three tables selected based on each coefficient's position, have parameter values that already compensate for factors of other transformation multiplications, except for those of a power of two, (e.g., two or one-half), which are performed by a shift operation during the transformation and inverse transformation processes. Computational complexity is significantly reduced with respect to other known transforms without adversely impacting compression or quality.

Block transform and quantization for image and video coding

The invention facilitates adaptive compression of multi-level images, such as captured digital images of a whiteboard, etc., encoding a bitstream comprising a color image component and a black-and-white image component. Either or both of a color and a black-and-white image can be output to a user based on user desires, receiving device capabilities, etc.

Adaptive compression of multi-level images

A warped spectral estimate of an original audio signal can be used to encode a representation of a fine estimate of the original signal. The representation of the warped spectral estimate and the representation of the fine estimate can be sent to a speech recognition system. The representation of the warped spectral estimate can be passed to a speech recognition engine, where it may be used for speech recognition. The representation of the warped spectral estimate can also be used along with the representation of the fine estimate to reconstruct a representation of the original audio signal.

Warped spectral and fine estimate audio encoding

A quality level determining the extent to which each image file is compressed is automatically computed for each image file in a set to ensure that the total size of the compressed image files does not exceed a predefined limit. The compressed size of each image file is initially determined when compressed at a predefined minimum acceptable level and at a nominal level. The relative complexity of the image files is determined based upon their high frequency energy content. As a function of the image file complexity, and starting with the compressed sizes initially determined, the appropriate quality level is determined for compressing each of the image files in an iterative process that ensures the total size of the compressed image files does not exceed the predefined limit, while retaining acceptable quality. Thus, a set of image files can be compressed optimally to fit within a limited storage.

Distributing limited storage among a collection of media objects

Harmonic distortion residual echo suppression (HDRES) technique embodiments are presented which act to suppress the residual echo remaining after a near-end microphone signal has undergone AEC, including harmonic distortion in the signal that was caused by the speaker audio signal playback. In general, an AEC module is employed which suppresses some parts of the speaker audio signal found in a near-end microphone signal and generates an AEC output signal. A HDRES module then inputs the AEC output signal and the speaker audio signal, and suppresses at least a portion of a residual part of the speaker audio signal that was left unsuppressed by the AEC module. This includes at least a portion of the harmonic distortion exhibited in the AEC output signal.

Harmonic distortion residual echo suppression

A system and method facilitating signal enhancement utilizing an adaptive filter is provided. The invention includes an adaptive filter that filters an input based upon a plurality of adaptive coefficients, the adaptive filter modifying at least one of the adaptive coefficients based on a feedback output. The invention further includes a feedback component that provides the feedback output based, at least in part, upon a non-linear function of the acoustic reverberation reduced output. The invention further provides a noise statistics component that stores noise statistics associated with a noise portion of an input signal and a signal+noise statistics component that stores signal+noise statistics associated with a signal and noise portion of the input signal. The invention further provides a spatial filter that provides an output signal based, at least in part, upon a filtered input signal, the filtering being based, at least in part, upon a weighted error calculation of the noise statistics and the signal+noise statistics.

Microphone array signal enhancement

An adaptive interpolation technique with artifact reduction is described that technique generates digital images with full-color RGB (red, green, blue) information, from raw pictures (e.g., Bayer-mosaiced single-color images) created by single-CCD digital cameras. The technique employs an improved criterion for choosing the interpolation criterion, which takes into account an output interpolated value. It employs small changes to filter coefficients, for better results and accommodation of 'correction attenuation'. In one embodiment, the technique further employs a 'correction attenuation' step, which reduces &#x201c;color sprinkling&#x201d; artifacts for certain kinds of diagonal edges. The technique makes only a single pass over the image; all colors are interpolated during that pass, vice the multiple passes required by other better performing algorithms (in some cases over ten).

Adaptive interpolation with artifact reduction of images

A 'STAC Codec' provides audio transcoding and decoding by processing an encoded audio signal using a backward-adaptive run-length Golomb-Rice (RLGR) decoder to recover transform coefficients of the encoded audio signal. The transform coefficients are then either transcoded in the transform domain to lossy or other formats, or decoded to the time domain by applying an inverse integer-reversible modulated lapped transform (MLT) to the recovered transform coefficients to recover an uncompressed time domain representation compressed audio signal. In additional embodiments, an inter-block spectral estimation and inverse data sorting strategy is used in recovering the transform coefficients from the encoded audio signal. In other embodiments, conversion from lossless encoding to near-lossless encoding is achieved by right-shifting recovered transform coefficients by some number of bits such that quantization errors are not perceived as distortion in the decoded audio signal, then re-encoding the right shifted transform coefficients.

Transform domain transcoding and decoding of audio data using integer-reversible modulated lapped transforms

Growing community of inventors

Sammamish, WA, United States of America

Henrique Sarmento Malvar