Conexant Systems, Inc.

Synaptics Corporation

Yair Kerner

Trausti Thormundsson

Ragnar Hlynur Jonsson

Harry K Lau

James Walter Wihardja

Jonathan Chien

Youhong Lu

Sverrir Olafsson

Govind Kannan

Mouna Elkhatib

Stuart Whitfield Hutson

Shlomi I Regev

Eitan David

Thorarinn Vikingur Sveinsson

Xiaoyan Vivian Qian

Bryan J Burns

Iddit Shalem

Randall Deetz

Systems and methods provide input and output mode control for audio processing on a user device. Audio processing may be configured by monitoring audio activity on a device having at least one microphone and a digital audio processing unit, collecting information from the monitoring of the activity, including an identification of at least one application utilizing audio processing, and determining a context for the audio processing, the context including at least one of a hardware, software, audio signal and/or environmental context. An audio signal processing configuration is determined based on the application and determined context, an associated audio signal processing mode is selected, and an optimized audio signal generated.

Input/output mode control for audio processing

A system for processing audio data comprising an amplifier configured to receive an audio signal and to perform nonlinear processing on the audio signal. An encoder coupled to the amplifier, the encoder configured to receive the nonlinearly processed audio signal and to encode the nonlinearly processed audio signal into a data transmission format. A transmitter configured to receive and transmit the encoded nonlinearly processed audio signal. A receiver configured to receive the transmitted encoded nonlinearly processed audio signal and to decode the encoded nonlinearly processed audio signal. A digital voice processor configured to receive the nonlinearly processed audio signal and to use the nonlinearly processed audio signal for echo estimation and to subsequently subtract the estimated echo signal from a microphone signal.

Class-D amplifier with pulse density modulation output feedback for higher performance acoustic echo canceller

A method for signal processing, receiving a time domain signal having a sample-rate Fs and generating N time domain signal bands, each having a bandwidth equal to Fs/N. Receiving the N signal bands and transforming a first time domain signal band to a frequency domain at a first resolution and a second time domain signal band to the frequency domain at a second resolution, where the first resolution may be different from the second resolution. Determining one or more first filter coefficients using the frequency domain components from the first signal band and one or more second filter coefficients using the frequency domain components from the second signal band. Transforming the first and second filter coefficients from the frequency domain to a time domain. Applying the first and second time domain filter coefficients to the first and second time domain signals, respectively.

Reduced-delay subband signal processing system and method

Traditionally, echo cancellation has employed linear adaptive filters to cancel echoes in a two way communication system. The rate of adaptation is often dynamic and varies over time. Disclosed are novel rates of adaptation that perform well in the presence of background noise, during double talk and with echo path changes. Additionally, the echo or residual echo can further be suppressed with non-linear processing performed using joint frequency-time domain processing.

Systems and methods for echo cancellation and echo suppression

Peak reduction and power limitations are used to prevent distortion and protect components. In a cellular telephone, peak reduction can be based on battery power level to prevent electrical distortion from saturation. In addition peak reduction can be used to prevent mechanical distortion such as rub and buzz. Dynamic range compression can be used for peak reduction. In another application dynamic range compression can be used to control the power output to protect a speaker from damage. One example of a dynamic range compressor/peak limiter comprises a look-ahead buffer and an analysis engine. For example, the look-ahead buffer holds a window of samples of a signal. The analysis engine selects a gain envelope function on the basis of the samples, for example, by selecting the Pth sample in the buffer whenever that sample exceeds a given threshold.

Waveform shaping system to prevent electrical and mechanical saturation in loud speakers

A sound reproduction apparatus comprising a loudspeaker bar having a plurality of loudspeakers. A loudspeaker bar controller coupled to the loudspeaker bar for processing audio data for the plurality of loudspeakers, the loudspeaker bar controller comprising a spatial enhancement/virtualization system for receiving a surround channel of audio data and processing the surround channel of audio data with a spatial generation/virtualization filter, wherein a left stereo channel of audio data and a right stereo channel of audio data are not processed with the head related transfer function filter. Bass is enhanced for small speakers which are not able to produce bass frequencies.

Systems and methods for speaker bar sound enhancement

There is provided a method for use in a computer for redirecting an audio stream from a first audio endpoint to a second audio endpoint in the computer having a controller and a computer operating system for execution by the controller. The method includes directing the audio stream from a client application running on the computer operating system through a first audio resource stack of the computer to the first audio endpoint. The method further includes creating an audio endpoint bridge using the computer operating system, where the audio endpoint bridge provides a path for the audio stream to be directed from the first audio resource stack to pass through a second audio resource stack to the second audio endpoint. The method may also enhance the audio stream to reduce noise and redirect the audio stream to the second audio endpoint using the audio endpoint bridge.

Method and system for removal of clicks and noise in a redirected audio stream

Disclosed methods and systems measure acoustic responses to training signals activated prior to communication sessions. Profiles related to the acoustic responses are saved and adapted during communication sessions. Training signals may have uniform frequency distributions over a frequency range and may be in response to user inputs, timeouts, or predetermined events. In the next excessive divergence is detected, an adapted profile may be substituted by an original, trained profile.

Acoustic echo canceller

An apparatus for providing real-time calibration for two or more microphones. A calibrator for receiving a left microphone signal and a right microphone signal and generating phase difference data. A phase and amplitude correction system for receiving one of the left microphone signal or the right microphone signal the phase difference data and generating calibration data for a beamformer. The beamformer receiving the calibration data, the left microphone signal and the right microphone signal and generating a monaural beamformed signal.

Microphone array calibration method and apparatus

A communication session is established between an APCM modem and a DPCM modem. The central office interrupts the communication session by transmitting a call-waiting alert signal to the APCM modem. In response to the alert signal, the APCM modem transmits a caller ID data request to the central office. The APCM modem further transmits an on hold notification to the DPCM modem to prevent the DPCM modem from terminating the communication session as a result of the interruption by the alert signal and/or receipt of the caller ID data by the APCM modem. The DPCM modem is placed on hold while the APCM modem answers the call waiting. Subsequently, the APCM modem initiates a quick connect procedure to re-establish the communication session with the DPCM modem.

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Kiryat Ono, Israel

Yair Kerner