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| 20080234845 | AUDIO COMPRESSION AND DECOMPRESSION USING INTEGER-REVERSIBLE MODULATED LAPPED TRANSFORMS - A “STAC Codec” provides lossless audio compression and decompression by processing an audio signal using integer-reversible modulated lapped transforms (MLT) to produce transform coefficients. Transform coefficients are then encoded using a backward-adaptive run-length Golomb-Rice (RLGR) encoder to produce losslessly compressed audio signals. In additional embodiments, further compression gains are achieved via an inter-block spectral estimation and data sorting strategy. Further, compression in the transform domain allows the bitstream to be partially decoded, using the corresponding RLGR decoder, to reconstruct the frequency-domain coefficients. These frequency-domain coefficients are then directly used to speed up various transform-domain based applications such as transcoding media to lossy or other formats, search, identification, visualization, watermarking, etc. In other embodiments, near-lossless compression is achieved by right-shifting transform coefficients by some number of bits such that quantization errors are not perceived as distortion in the decoded audio signal. | 09-25-2008 |
| 20080234846 | TRANSFORM DOMAIN TRANSCODING AND DECODING OF AUDIO DATA USING INTEGER-REVERSIBLE MODULATED LAPPED TRANSFORMS - 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. | 09-25-2008 |
| 20080240559 | ADAPTIVE INTERPOLATION WITH ARTIFACT REDUCTION OF IMAGES - 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 “color sprinkling” 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). | 10-02-2008 |
| 20080243497 | STATIONARY-TONES INTERFERENCE CANCELLATION - An “Interference Canceller” provides a computationally efficient real-time technique for removing stationary-tone interference from signals. Typical sources of stationary tone contamination of signals include noise from power wiring (i.e., 50/60 Hz or 400 Hz and their harmonics), frame or line frequencies from electronic devices, and noise from computer fans, hard disk drives, etc. In general, the Interference Canceller adaptively builds and updates a model of stationary tone interference in consecutive frames of an input signal. This adaptively updated model is then used to extrapolate and subtract noise from subsequent frames of the input signal to generate a “clean” output signal. This output signal exhibits significant attenuation of stationary tone interference without eliminating important portions of the underlying signal or distorting the underlying signal with artifacts such as musical noise or nonlinear distortions. The Interference Canceller is applicable for use either alone, or as pre-processor to conventional noise suppression. | 10-02-2008 |
| 20090214048 | HARMONIC DISTORTION RESIDUAL ECHO SUPPRESSION - 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. | 08-27-2009 |
| 20090319278 | EFFICIENT CODING OF OVERCOMPLETE REPRESENTATIONS OF AUDIO USING THE MODULATED COMPLEX LAPPED TRANSFORM (MCLT) - 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. | 12-24-2009 |
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| 20090238475 | DISTRIBUTING LIMITED STORAGE AMONG A COLLECTION OF MEDIA OBJECTS - 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. | 09-24-2009 |
| 20110116543 | BLOCK TRANSFORM AND QUANTIZATION FOR IMAGE AND VIDEO CODING - 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. | 05-19-2011 |
