# US 2019/0172450 A1 · Voice enhancement in audio signals through modified generalized eigenvalue beamformer > Agents: this data is available as JSON from the patentgrep API and MCP server — see https://patentgrep.com/llms.txt - Number: US 2019/0172450 A1 - Document ID: US20190172450A1 - Status: Published application - Kind: A1 - Application number: 15/833,977 - Filed: 2017-12-06 - Published: 2019-06-06 - Earliest claimed: 2017-12-06 - Assignees: Synaptics Incorporated - Applicants: Synaptics Incorporated - Inventors: Frederic Philippe Denis Mustiere; Francesco Nesta - Web page: https://patentgrep.com/patent/US20190172450A1 ## CPC - G10L 15/20: Speech recognition techniques specially adapted for robustness in adverse environments, e.g. in noise, of stress induced speech (G10L21/02 takes precedence) - G10L 15/22: Procedures used during a speech recognition process, e.g. man-machine dialogue - G10L 21/0232: Processing in the frequency domain - G10L 21/038: using band spreading techniques - G10L 25/21: the extracted parameters being power information - G10L 25/84: for discriminating voice from noise - H04R 3/005: {for combining the signals of two or more microphones (specially adapted for hearing aids H04R25/407)} - H04R 27/00: Public address systems (circuits for preventing acoustic reaction H04R3/02; circuits for distributing signals to loudspeakers H04R3/12; {monitoring or testing arrangements for public address systems H04R29/007}; amplifiers H03F) - G10L 25/78: Detection of presence or absence of voice signals (switching of direction of transmission by voice frequency in two-way loud-speaking telephone systems H04M9/10) - G10L 2015/223: {Execution procedure of a spoken command} - G10L 2021/02166: {Microphone arrays; Beamforming} - H04R 1/406: {microphones} - H04R 5/02: Spatial or constructional arrangements of loudspeakers - H04R 5/04: Circuit arrangements, {e.g. for selective connection of amplifier inputs/outputs to loudspeakers, for loudspeaker detection, or for adaptation of settings to personal preferences or hearing impairments (combinations of amplifiers H03F3/68; stereophonic systems H04S)} - H04R 2227/001: Adaptation of signal processing in PA systems in dependence of presence of noise - H04R 2227/003: Digital PA systems using, e.g. LAN or internet - H04R 2227/005: Audio distribution systems for home, i.e. multi-room use - H04R 2227/009: Signal processing in [PA] systems to enhance the speech intelligibility - G10L: SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING - H04R: LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS (producing sounds with frequency not determined by supply frequency G10K) ## Abstract A real-time audio signal processing system includes an audio signal processor configured to process audio signals using a modified generalized eigenvalue (GEV) beamforming technique to generate an enhanced target audio output signal. The digital signal processor includes a sub-band decomposition circuitry configured to decompose the audio signal into sub-band frames in the frequency domain and a target activity detector configured to detect whether a target audio is present in the sub-band frames. Based on information related to the sub-band frames and the determination of whether the target audio is present in the sub-band frames, the digital signal processor is configured to use the modified GEV technique to estimate the relative transfer function (RTF) of the target audio source, and generate a filter based on the estimated RTF. The filter may then be applied to the audio signals to generate the enhanced audio output signal. ## Claims ### Claim 1 A method for processing audio signals, comprising: receiving a multichannel audio signal based on audio inputs detected by a plurality of audio input components; determining whether the multichannel audio signal comprises target audio associated with an audio source; estimating a relative transfer function of the audio source with respect to the plurality of audio input components based on the multichannel audio signal and a determination of whether the multichannel audio signal comprises the target audio; and processing the multichannel audio signal to generate an audio output signal by enhancing the target audio in the multichannel audio signal based on the estimated relative transfer function. ### Claim 2 (depends on 1) The method of claim 1, further comprising transforming the multichannel audio signal to sub-band frames according to a plurality of frequency sub-bands, wherein the estimating the relative transfer function of the audio source is further based on the sub-band frames. ### Claim 3 (depends on 1) The method of claim 1, wherein the estimating the RTF comprises computing a vector. ### Claim 4 (depends on 3) The method of claim 3, further comprising: generating a noise power spectral density matrix representing characteristics of noise in the audio inputs; and inverting the noise power spectral density matrix to generate an inverted noise power spectral density matrix, wherein the computing the vector comprises using a single function to directly update the inverted noise power spectral density matrix based on the audio signal in response to a determination that the multichannel audio signal does not comprise the target audio. ### Claim 5 (depends on 3) The method of claim 3, further comprising generating a target audio power spectral density matrix representing characteristics of the target audio in the audio inputs, wherein the computing the vector comprises updating the target audio power spectral density matrix based on the multichannel audio signal in response to a determination that the multichannel audio signal comprises the target audio. ### Claim 6 (depends on 3) The method of claim 3, wherein the computed vector is an eigenvector. ### Claim 7 (depends on 3) The method of claim 3, wherein the computing the vector comprises using an iterative extraction algorithm to compute the vector. ### Claim 8 (depends on 1) The method of claim 1, wherein the plurality of audio input components comprises an array of microphones. ### Claim 9 (depends on 8) The method of claim 8, further comprising outputting the audio output signal. ### Claim 10 (depends on 9) The method of claim 9, wherein the audio output signal is output to an external device over a network. ### Claim 11 (depends on 8) The method of claim 8, further comprising: determining a command based on the audio output signal; and transmitting the command to an external device. ### Claim 12 (depends on 11) The method of claim 11, further comprising: receiving data from the external device based on the transmitted command; and in response to receiving the data from the external device, providing an output via one or more speakers based on the received data. ### Claim 13 An audio processing device comprising: a plurality of audio input components configured to detect audio inputs and generate a multichannel audio signal based on the detected audio inputs; an activity detector configured to determine whether the multichannel audio signal comprises target audio associated with an audio source; a target audio source RTF estimator configured to estimate the relative transfer function of the audio source with respect to the plurality of audio input components based on the multichannel audio signal and a determination of whether the multichannel audio signal comprises the target audio; and an audio signal processor configured to process the multichannel audio signal to generate an audio output signal by enhancing the target audio signal in the multichannel audio signal based on the estimated relative transfer function. ### Claim 14 (depends on 13) The audio processing device of claim 13, further comprising: a sub-band decomposer configured to transform the multichannel audio signal to sub-band frames according to a plurality of frequency sub-bands, wherein the audio source RTF estimator is configured to estimate the relative transfer function of the audio source based further on the sub-band frames. ### Claim 15 (depends on 13) The audio processing device of claim 13, wherein the audio source RTF estimator is configured to estimate the relative transfer function of the audio source by computing a vector. ### Claim 16 (depends on 15) The audio processing device of claim 15, wherein the audio source RTF estimator is configured to compute the vector for the audio signal by: generating a noise power spectral density matrix representing characteristics of noise in the audio inputs; and inverting the noise power spectral density matrix to generate an inverted noise power spectral density matrix, wherein the computing the vector comprises using a single function to directly update the inverted noise power spectral density matrix based on the multichannel audio signal. ### Claim 17 (depends on 15) The audio processing device of claim 15, wherein the vector is computed using an iterative extraction algorithm. ### Claim 18 (depends on 13) The audio processing device of claim 13, wherein the plurality of audio input components comprises an array of microphones. ### Claim 19 (depends on 13) The audio processing device of claim 13, further comprising one or more speakers configured to output the audio output signal. ### Claim 20 (depends on 13) The audio processing device of claim 13, further comprising a network interface configured to transmit the audio output signal to an external device. ### Claim 21 (depends on 13) The audio processing device of claim 13, further comprising a speech recognition engine configured to determine one or more words based on the audio output signal. ### Claim 22 (depends on 21) The audio processing device of claim 21, wherein the speech recognition engine is further configured to map the one or more words to a command. ### Claim 23 (depends on 13) The audio processing device of claim 13, wherein the desired audio signal comprises a voice signal, and wherein the activity detector is a voice activity detector.