An audio processing method may involve receiving output signals from each microphone of a plurality of microphones in an audio environment, the output signals corresponding to a current utterance of a person and determining, based on the output signals, one or more aspects of context information relating to the person, including an estimated current proximity of the person to one or more microphone locations. The method may involve selecting two or more loudspeaker-equipped audio devices based, at least in part, on the one or more aspects of the context information, determining one or more types of audio processing changes to apply to audio data being rendered to loudspeaker feed signals for the audio devices and causing one or more types of audio processing changes to be applied. In some examples, the audio processing changes have the effect of increasing a speech to echo ratio at one or more microphones.

A method for detecting and flagging problems in videoconferencing is provided. The method includes measuring, by a processing device of a user videoconferencing device, a first set of acoustic data associated with a conference room during a video conference attended by a plurality of users. The first set of acoustic data indicates one or more acoustic parameters, including the reverberation time, the signal-to-noise ratio, or the absolute noise level of the conference room. The method further includes comparing the one or more first acoustic parameters to one or more thresholds during the video conference. Moreover, the method includes providing at least one of the first acoustic parameters or a result of the comparison to the at least one user during the video conference.

An audio device is provided. The audio device includes processing circuitry which is connected to a loudspeaker and a microphone. The processing circuitry is configured to play an echo reference signal from a far end on the loudspeaker, and perform an acoustic echo cancellation (AEC) process using the echo reference signal and an acoustic signal received by the microphone using an AEC adaptive filter. The processing circuitry repeatedly determines a first status of the loudspeaker according to a relation between the played echo reference signal and the received acoustic signal, and transmits a first status signal indicating the first status of the loudspeaker to the far end through a cloud network.

This disclosure describes an apparatus and method of an embodiment of an invention that is a conferencing device that includes a microphone array that further comprises a plurality of microphones; a processor configured to execute the following steps: performing a beamforming operation to combine the plurality of microphones into a plurality of combined signals that is greater in number than one and less in number than the plurality of microphone signals, each of the plurality of combined signals corresponding to a different fixed beam; wherein performing the beamforming operation includes applying beamforming weights to the signals from each microphone to achieve a desired pickup pattern that includes a main lobe and sidelobes together with nulls for each fixed beam; performing an acoustic echo cancellation operation generate a plurality of combined echo cancelled signals; and selecting one or more of the combined echo cancelled signals for transmission to the far end.

A head mounted display (HMD) worn on a head of a first person that provides an electronic call between the first person and a second person. The HMD includes or more sensors that determine a size of a physical environment where the first person is located and objects in the physical environment. A display displays a virtual image of the second person during the electronic call, and one or more processors adjust a voice of the second person played to the first person to compensation for sound reverberation. The HMD include speakers that play the voice of the second person in binaural sound compensated for the sound reverberation.

A playback device is configured to: produce a first channel audio output of a first channel of audio content; produce a second channel audio output of a second channel of the audio content; receive captured audio content comprising (i) a first portion corresponding to the first channel audio output, (ii) a second portion corresponding to the second channel audio output, and (iii) a third portion corresponding to a voice command, wherein the captured audio content has a first signal-to-noise ratio; determine a set of signal components from at least one of the first channel or the second channel of the audio content; perform acoustic echo cancellation on a subset of signal components; determine an acoustic echo cancellation output; and apply the acoustic echo cancellation output to the captured audio content and thereby increase the first signal-to-noise ratio to a second signal-to-noise ratio that is greater than the first signal-to-noise ratio.

Some implementations involve receiving a content stream that includes audio data, determining a content type corresponding to the content stream and determining, based at least in part on the content type, a noise compensation method. Some examples involve performing the noise compensation method on the audio data to produce noise-compensated audio data, rendering the noise-compensated audio data for reproduction via a set of audio reproduction transducers of the audio environment, to produce rendered audio signals, and providing the rendered audio signals to at least some audio reproduction transducers of the audio environment.

An electronic device according to various embodiments comprises a communication module including communication circuitry, at least one microphone, and a processor. The processor can be configured to: be connected to an external device through the communication module; switch a call audio signal path to an external device output path in response to being call-connected to an opponent device; duplicate a call audio signal (Rx in) transmitted from the opponent device, provide a first signal to an echo removal module, and provide a second signal to the external device via the external device output path; measure latency between the external device and the electronic device; variably adjust the size of a dynamic buffer for removing echo, by applying the measured latency; generate a reference signal delayed in the second signal, by the adjusted dynamic buffer; and remove an echo signal, with respect to the first signal generated from a speaker of the external device, from a microphone input signal obtained from the at least one microphone, based on the generated reference signal.

Embodiments of the present application relate to a dual-microphone array echo cancellation method, device and electronic equipment, comprises: the acquired distal signal, the first proximal signal and the second proximal signal are processed by linear filtering; obtain initial suppression gain factors and variable step-size factors of the first near-end signal and the second near-end signal; performing residual echo suppression on the first error frequency spectrum and the second error frequency spectrum by using an adaptive zero-pole echo canceller to obtain a target frequency spectrum signal; performing sub-band range selection on the initial suppression gain factors to obtain a smooth factor, and performing full-band smoothing and exponential operation to obtain a secondary suppression gain factor; performing filtering processing on the target frequency spectrum signal by using the secondary suppression gain factor to obtain a target near-end voice signal.

A double-talk state detection method includes: calculating an energy ratio between a first energy of an error signal in each sub-band of M sub-bands and a second energy of a filtered signal in the same sub-band as the error signal, thereby obtaining M energy ratios, where the error signal is a difference between an input signal collected by a microphone and the filtered signal, the filtered signal is a signal obtained after performing filtering process on a reference signal, and M is a positive integer; performing a first smoothing processing on the M energy ratios to obtain M first energy smoothing ratios, and performing a second smoothing processing on the M first energy smoothing ratios to obtain M second energy smoothing ratios; performing double-talk state detection based on the M first energy smoothing ratios and the M second energy smoothing ratios to determine a state of the input signal.