Abstract: 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.

Abstract: A system and process for estimating the time delay of arrival (TDOA) between a pair of audio sensors of a microphone array is presented. Generally, a generalized cross-correlation (GCC) technique is employed. However, this technique is improved to include provisions for both reducing the influence (including interference) from correlated ambient noise and reverberation noise in the sensor signals prior to computing the TDOA estimate. Two unique correlated ambient noise reduction procedures are also proposed. One involves the application of Wiener filtering, and the other a combination of Wiener filtering with a Gnn subtraction technique. In addition, two unique reverberation noise reduction procedures are proposed. Both involve applying a weighting factor to the signals prior to computing the TDOA which combines the effects of a traditional maximum likelihood (TML) weighting function and a phase transformation (PHAT) weighting function.

Abstract: Systems and methods for performing adaptive filtering are disclosed. The present invention generates probabilities that can be used in an encoder, such as an arithmetic encoder and generates those probabilities in a computationally efficient manner. Probabilities of previously encoded coefficients are employed, effectively, in generating probabilities of the coefficients without regard to directional information. Thus, a large amount of information is adaptively and efficiently used in generating the probabilities. For the coefficients, the probability is computed based at least partly on at least one probability of a previously computed probability of a neighboring coefficient. Then, the coefficients are encoded using those computed probabilities.

Abstract: A system and process for finding the location of a sound source using direct approaches having weighting factors that mitigate the effect of both correlated and reverberation noise is presented. When more than two microphones are used, the traditional time-delay-of-arrival (TDOA) based sound source localization (SSL) approach involves two steps. The first step computes TDOA for each microphone pair, and the second step combines these estimates. This two-step process discards relevant information in the first step, thus degrading the SSL accuracy and robustness. In the present invention, direct, one-step, approaches are employed. Namely, a one-step TDOA SSL approach and a steered beam (SB) SSL approach are employed. Each of these approaches provides an accuracy and robustness not available with the traditional two-step approaches.

Abstract: A system and process for estimating the time delay of arrival (TDOA) between a pair of audio sensors of a microphone array is presented. Generally, a generalized cross-correlation (GCC) technique is employed. However, this technique is improved to include provisions for both reducing the influence (including interference) from correlated ambient noise and reverberation noise in the sensor signals prior to computing the TDOA estimate. Two unique correlated ambient noise reduction procedures are also proposed. One involves the application of Wiener filtering, and the other a combination of Wiener filtering with a Gnn subtraction technique. In addition, two unique reverberation noise reduction procedures are proposed. Both involve applying a weighting factor to the signals prior to computing the TDOA which combines the effects of a traditional maximum likelihood (TML) weighting function and a phase transformation (PHAT) weighting function.

Abstract: An “adaptive audio playback controller” operates by decoding and reading received packets of an audio signal into a signal buffer. Samples of the decoded audio signal are then played out of the signal buffer according to the needs of a player device. Jitter control and packet loss concealment are accomplished by continuously analyzing buffer content in real-time, and determining whether to provide unmodified playback from the buffer contents, whether to compress buffer content, stretch buffer content, or whether to provide for packet loss concealment for overly delayed or lost packets as a function of buffer content. Further, the adaptive audio playback controller also determines where to stretch or compress particular frames or signal segments in the signal buffer, and how much to stretch or compress such segments in order to optimize perceived playback quality.

Abstract: An adaptive “temporal audio scaler” is provided for automatically stretching and compressing frames of audio signals received across a packet-based network. Prior to stretching or compressing segments of a current frame, the temporal audio scaler first computes a pitch period for each frame for sizing signal templates used for matching operations in stretching and compressing segments. Further, the temporal audio scaler also determines the type or types of segments comprising each frame. These segment types include “voiced” segments, “unvoiced” segments, and “mixed” segments which include both voiced and unvoiced portions. The stretching or compression methods applied to segments of each frame are then dependent upon the type of segments comprising each frame. Further, the amount of stretching and compression applied to particular segments is automatically variable for minimizing signal artifacts while still ensuring that an overall target stretching or compression ratio is maintained for each frame.

Abstract: A “speech onset detector” provides a variable length frame buffer in combination with either variable transmission rate or temporal speech compression for buffered signal frames. The variable length buffer buffers frames that are not clearly identified as either speech or non-speech frames during an initial analysis. Buffering of signal frames continues until a current frame is identified as either speech or non-speech. If the current frame is identified as non-speech, buffered frames are encoded as non-speech frames. However, if the current frame is identified as a speech frame, buffered frames are searched for the actual onset point of the speech. Once that onset point is identified, the signal is either transmitted in a burst, or a time-scale modification of the buffered signal is applied for compressing buffered frames beginning with the frame in which onset point is detected. The compressed frames are then encoded as one or more speech frames.

Abstract: A system and process for finding the location of a sound source using direct approaches having weighting factors that mitigate the effect of both correlated and reverberation noise is presented. When more than two microphones are used, the traditional time-delay-of-arrival (TDOA) based sound source localization (SSL) approach involves two steps. The first step computes TDOA for each microphone pair, and the second step combines these estimates. This two-step process discards relevant information in the first step, thus degrading the SSL accuracy and robustness. In the present invention, direct, one-step, approaches are employed. Namely, a one-step TDOA SSL approach and a steered beam (SB) SSL approach are employed. Each of these approaches provides an accuracy and robustness not available with the traditional two-step approaches.

Abstract: A system and method for embedding information into digital media and later detecting the embedded information using a unique spread spectrum modulation technique. In general, the present invention removes interference caused by an original signal from the detection process thereby eliminating a major source of detection error. The interference caused by the original signal is removed by using the encoder knowledge about the original signal and modulating the energy of the embedded mark to compensate for the original signal interference. The present invention also includes a novel redundant bit representation technique causes a resulting average over a large sample to tend to zero, thereby reducing the vulnerability of the present invention to malicious collusion attacks.

Abstract: Various embodiments reduce noise within a particular environment, while isolating and capturing speech in a manner that allows operation within an otherwise noisy environment. In one embodiment, an array of one or more microphones is used to selectively eliminate noise emanating from known, generally fixed locations, and pass signals from a pre-specified region or regions with reduced distortion.

Abstract: A system and process for estimating the time delay of arrival (TDOA) between a pair of audio sensors of a microphone array is presented. Generally, a generalized cross-correlation (GCC) technique is employed. However, this technique is improved to include provisions for both reducing the influence (including interference) from correlated ambient noise and reverberation noise in the sensor signals prior to computing the TDOA estimate. Two unique correlated ambient noise reduction procedures are also proposed. One involves the application of Wiener filtering, and the other a combination of Wiener filtering with a Gnn subtraction technique. In addition, two unique reverberation noise reduction procedures are proposed. Both involve applying a weighting factor to the signals prior to computing the TDOA which combines the effects of a traditional maximum likelihood (TML) weighting function and a phase transformation (PHAT) weighting function.

Abstract: A system and method for embedding information into digital media and later detecting the embedded information using a unique spread spectrum modulation technique. In general, the present invention removes interference caused by an original signal from the detection process thereby eliminating a major source of detection error. The interference caused by the original signal is removed by using the encoder knowledge about the original signal and modulating the energy of the embedded mark to compensate for the original signal interference. The present invention also includes a novel redundant bit representation technique causes a resulting average over a large sample to tend to zero, thereby reducing the vulnerability of the present invention to malicious collusion attacks.

Abstract: An apparatus and method for changing the bitrate of an encoded bitstream in a compressed domain, i.e., a domain where the input image is still represented by transform coefficients, e.g., DCT coefficients or wavelet coefficients is disclosed. The bitrate of the encoded bitstream is changed by applying a different quantization scale directly to the transform coefficients of the encoded bitstream. The requantization error or distortion introduced by the requantization process is addressed by propagating the requantization error to the following frame.

Abstract: A conventional variable-length codebook is converted into a modified codebook in which each symbol is represented by the same number of bits, but at least one symbol can be represented by at least two different code words. Such symbols therefore may be said to have redundant bits, which may then be used to encode other symbols, thereby achieving a degree of data compression. By using fixed-length code words, the positions of code words in the resulting encoded data stream are known a priori. As a result, the encoded data can decoded in parallel. The present invention provides both the data compression advantages of variable-length encoding schemes and the ability to perform parallel decoding processing of fixed-length encoding schemes. The present invention can be embodied in either lossless or lossy implementations.

Abstract: A method and apparatus that inserts watermark information directly into an encoded video bitstream. The method identifies specific blocks or macroblocks in an encoded video bitstieam and inserts the watermark information directly into the bitstream such that these selected blocks are replaced with a block containing watermark information or augmented with watermark information.