Google Inc.

Lsi Logic Corporation

Lsi Corporation

Joseph Caroselli

Arun Narayanan

Richard Cameron Rose

Izhak Shafran

Charles E MacDonald

Shirish A Altekar

Ye Liu

Jenn-Gang Chern

Yiteng Huang

Cathy Ye Liu

Vishnu Balan

Chintan M Desai

Implementations disclosed herein are directed to initializing and utilizing a beamformer in processing of audio data received at a computing device. The computing device can: receive audio data that captures a spoken utterance of a user, determine that a first audio data segment of the audio data includes one or more particular words or phrases; obtain a preceding audio data segment that precedes the first audio data segment; estimate a spatial correlation matrix based on the first audio data segment and based on the preceding audio data segment; initialize the beamformer based on the estimated spatial correlation matrix; and cause the initialized beamformer to be utilized in processing of at least a second audio data segment of the audio data. Additionally, or alternatively, the computing device can transmit the spatial correlation matrix to server(s), and the server(s) can transmit the initialized beamformer back to the computing device.

Context aware beamforming of audio data

Utilizing an adaptive multichannel technique to mitigate reverberation present in received audio signals, prior to providing corresponding audio data to one or more additional component(s), such as automatic speech recognition (ASR) components. Implementations disclosed herein are 'adaptive', in that they utilize a filter, in the reverberation mitigation, that is online, causal and varies depending on characteristics of the input. Implementations disclosed herein are 'multichannel', in that a corresponding audio signal is received from each of multiple audio transducers (also referred to herein as &#x201c;microphones&#x201d;) of a client device, and the multiple audio signals (e.g., frequency domain representations thereof) are utilized in updating of the filter&#x2014;and dereverberation occurs for audio data corresponding to each of the audio signals (e.g., frequency domain representations thereof) prior to the audio data being provided to ASR component(s) and/or other component(s).

Adaptive multichannel dereverberation for automatic speech recognition

A multi-phase adaptive decision feedback equalizer minimizes post-cursor inter-symbol interference in a current data bit based on values of subsequent data bits in a data communication system. In one form, the receiver includes a plurality of modules each having a respective adaptive decision feedback equalizer. A processor responsive to output signals from each of the plurality of modules generates a plurality of coefficient values. The adaptive decision feedback equalizer has a plurality of taps receiving a respective output signal from one of the modules and a respective coefficient value to generate a respective correction signal. The correction signals are summed with the data signal and processed to recover the data. Pre-calculation of coefficients permits rapid selection of data. Multi-phase operation permits higher data frequencies.

Serial data link using decision feedback equalization

The present invention is directed to a detector for a high-density magnetic recording channel and other partial response channels. The present invention presents a method for decoding a high rate product code and a decoder which uses this method, comprising receiving a high rate product code; using a row detector to find a most likely codeword and a most likely error sequence for each row; and using a column processor to correct any remaining errors based on column parity bits and the most likely error sequence of each row. In a first aspect of the present invention, the row detector is implemented through a 2-VA detector. In a second aspect of the present invention, the row detector is implemented through a conventional VA detector and a hank of matched filters.

High rate product code decoding for partial response channel

The present invention describes a system and method for encoding a sequence of 64 bit digital data words into a sequence of 65 bit codewords having constraints of (d&equals;0, G&equals;11/I&equals;10) for recording upon a magnetic medium within a magnetic recording channel are disclosed. The method for encoding a sequence of 64 bit digital data words into a sequence of codewords having 65 bits, comprising the steps of dividing each 64-bit digital data word into 8-bit bytes, encoding two 8-bit bytes to form a 17-bit word, forming five 11-bit intermediate blocks from the 8-bit bytes, encoding the five 11-bit intermediate blocks, and concatenating the five encoded 11-bit intermediate blocks and uncoded and unconstrained bits from the 64 bit digital data word to form a 65 bit codeword. A corresponding decoding method is also described. A byte shuffler may be used in the processing.

Rate 64/65 (d&equals;0, G&equals;11/I&equals;10) run length limited modulation code

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Palo Alto, CA, United States of America

Joseph Caroselli