Qualcomm Incorporated

Asif Mohammad

Erik Visser

Lae-Hoon Kim

Samir Kumar Gupta

Ian Ernan Liu

Shaun William Van Dyken

Sreekanth Narayanaswamy

Dinesh Ramakrishnan

Deepak Kumar Challa

Heejong Yoo

Babak Forutanpour

Kwokleung Chan

Eddie L T Choy

Louis Dominic Oliveira

Wade Lyle Heimbigner

Jongwon Shin

Ye Jiang

Phuong Lam Ton

Andre Schevciw

Catalin Lacatus

Troy Schultz

Steven L Grant

Rishabh Tyagi

A vehicle includes an interface device, an in-vehicle control unit, a functional unit, and a processing circuitry. The interface device receives a spoken command to identify an in-cabin vehicle zone of two or more in-cabin vehicle zones of the vehicle, and receives background audio data concurrently with a portion of the spoken command. The in-cabin vehicle control unit separates the background audio data from the spoken command, and selects which in-cabin vehicle zone of the two or more in-cabin vehicle zones is identified by the spoken command. The functional unit controls a function within the vehicle. The processing circuitry stores, to a command buffer, data processed from the received spoken command, and controls, based on the data processed from the received spoken command, the functional unit using audio input received from the selected in-cabin vehicle zone.

In-vehicle voice command control

A headphone device includes a speaker housing, a first feedforward microphone, and a second feedforward microphone. The first feedforward microphone is coupled to the speaker housing at a first location, and the second feedforward microphone is coupled to the speaker housing at a second location. The headphone device also includes an active noise cancelling (ANC) circuit configured to generate an anti-noise signal based on at least one of a first signal from the first feedforward microphone and a second signal from the second feedforward microphone. The headphone device also includes a speaker configured to generate an audio output at least partially based on the anti-noise signal.

Directional noise cancelling headset with multiple feedforward microphones

An apparatus includes multiple microphones to generate audio signals based on sound of a far-field acoustic environment. The apparatus also includes a signal processing system to process the audio signals to generate at least one processed audio signal. The signal processing system is configured to update one or more processing parameters while operating in a first operational mode and is configured to use a static version of the one or more processing parameters while operating in the second operational mode. The apparatus further includes a keyword detection system to perform keyword detection based on the at least one processed audio signal to determine whether the sound includes an utterance corresponding to a keyword and, based on a result of the keyword detection, to send a control signal to the signal processing system to change an operational mode of the signal processing system.

Directional processing of far-field audio

A system includes a memory configured to store data associated with a service that is available. The system also includes a microphone associated with an acoustic space and configured to receive an audio input produced by a person. The system further includes a sensor located within the acoustic space and configured to detect vibrations produced by the person. The system includes a processor coupled to the memory, to the microphone, and to the sensor. The processor is configured to conditionally authorize execution of the service requested by the person, the service conditionally authorized based on the audio input and the vibrations.

Identification and authentication in a shared acoustic space

A system includes a feedback filter configured to receive an audio signal from a microphone and to process the audio signal based on a feedback path between a loudspeaker and the microphone. The system also includes an adaptive filter coupled to the feedback filter. The system further includes a signal combiner coupled to the microphone and to the adaptive filter.

In-vehicle communication signal processing

A multichannel acoustic system (MAS) comprises an arrangement of microphones and loudspeakers and a multichannel acoustic processor (MAP) to together enhance conversational speech between two or more persons in a shared acoustic space such as an automobile. The enhancements are achieved by receiving sound signals substantially originating from relatively near sound sources; filtering the sound signals to cancel at least one echo signal detected for at least one microphone from among the plurality of microphones; filtering the sound signals received by the plurality of microphones to cancel at least one feedback signal detected for at least one microphone from among the plurality of microphones; and reproducing the filtered sound signals for each microphone from among the plurality of microphones on a subset of loudspeakers corresponding that are relatively far from the source microphone.

Enhanced conversational communications in shared acoustic space

A method for multi-channel echo cancellation and noise suppression is described. One of multiple echo estimates is selected for non-linear echo cancellation. Echo notch masking is performed on a noise-suppressed signal based on an echo direction of arrival (DOA) to produce an echo-suppressed signal. Non-linear echo cancellation is performed on the echo-suppressed signal based, at least in part, on the selected echo estimate.

Multi-channel echo cancellation and noise suppression

A multi-channel sound (MCS) system features intelligent calibration (e.g., of acoustic echo cancellation (AEC)) for use in dynamic acoustic environments. A sensor subsystem is utilized to detect and identify changes in the acoustic environment and determine a 'scene' corresponding to the resulting acoustic characteristics for that environment. This detected scene is compared to predetermined scenes corresponding to the acoustic environment. Each predetermined scene has a corresponding pre-tuned filter configuration for optimal AEC performance. Based on the results of the comparison, the pre-tuned filter configuration corresponding to the predetermined scene that most closely matches the detected scene is utilized by the AEC subsystem of the multi-channel sound system.

Calibration of acoustic echo cancelation for multi-channel sound in dynamic acoustic environments

A multichannel acoustic system (MAS) comprises an arrangement of microphones, loudspeakers, and filters along with a multichannel acoustic processor (MAP) and other components to together provide and enhance the auditory experience of persons in a shared acoustic space such as, for example, the driver and other passengers in an automobile. Driver-specific features such as navigation and auditory feedback cues are described, as individual auditory customizations and collective communications both within the shared acoustic space as well as with other individuals not located in the space via enhanced conference call facilities.

Enhanced auditory experience in shared acoustic space

A crosstalk cancelation technique reduces feedback in a shared acoustic space by canceling out some or all parts of sound signals that would otherwise be produced by a loudspeaker to only be captured by a microphone that, recursively, would cause these sounds signals to be reproduced again on the loudspeaker as feedback. Crosstalk cancelation can be used in a multichannel acoustic system (MAS) comprising an arrangement of microphones, loudspeakers, and a processor to together enhance conversational speech between in a shared acoustic space. To achieve crosstalk cancelation, a processor analyzes the inputs of each microphone, compares it to the output of far loudspeaker(s) relative to each such microphone, and cancels out any portion of a sound signal received by the microphone that matches signals that were just produced by the far loudspeaker(s) and sending only the remaining sound signal (if any) to such far loudspeakers.

Feedback cancelation for enhanced conversational communications in shared acoustic space

A method for echo reduction by an electronic device is described. The method includes nulling at least one speaker. The method also includes mixing a set of runtime audio signals based on a set of acoustic paths to determine a reference signal. The method also includes receiving at least one composite audio signal that is based on the set of runtime audio signals. The method further includes reducing echo in the at least one composite audio signal based on the reference signal.

Systems and methods for surround sound echo reduction

Apparatus and methods for audio noise attenuation are disclosed. An audio signal analyzer can determine whether an input audio signal received from a microphone device includes a noise signal having identifiable content. If there is a noise signal having identifiable content, a content source is accessed to obtain a copy of the noise signal. An audio canceller can generate a processed audio signal, having an attenuated noise signal, based on comparing the copy of the noise signal to the input audio signal. Additionally or alternatively, data may be communicated on a communication channel to a separate media device to receive at least a portion of the copy of the noise signal from the separate media device, or to receive content-identification data corresponding to the content source.

Content based noise suppression

A method for echo cancellation and noise suppression is disclosed. Linear echo cancellation (LEC) is performed for a primary microphone channel on an entire frequency band or in a range of frequencies where echo is audible. LEC is performed on one or more secondary microphone channels only on a lower frequency range over which spatial processing is effective. The microphone channels are spatially processed over the lower frequency range after LEC. Non-linear noise suppression post-processing is performed on the entire frequency band. Echo post-processing is performed on the entire frequency band.

Integrated echo cancellation and noise suppression

An echo canceller for improved recognition and removal of an echo from a communication device. The echo canceller can dynamically reduce echo using an improved energy estimator and an improved adaptive filter. The improved energy estimator can determine if conversation is in a single talk period or a double talk period based on the combined energy of both the near end background noise and speech. The improved adaptive filter can reduce echo by dynamically changing adaptation speed or step size. In double talk, the adaptive filter(s) can dynamically slow-down or stop adaptation. In single talk, the filter can dynamically increase the speed of adaptation to improve accuracy, or decrease adaptation speed for stability.

System and method for echo cancellation

A communications device that is configured to detect double talk is described. An echo canceller is configured to cancel an echo from an input signal using an adaptive filter. A double-talk detector provides a double-talk statistic. The double-talk statistic is proportional to the ratio of the remaining echo energy in the cancellation error signal and the total cancellation error energy.

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San Diego, CA, United States of America

Asif Mohammad