A hands-free audio device has a receive path and a transmit path, which may operate at different audio sampling rates. The transmit path has an interference suppressor that receives a reference signal from the receive path and that suppresses interference in microphone signals received from a microphone array. The interference suppressor is followed in the transmit path by a multi-channel adaptive beamformer that produce a plurality of directional audio signals. A beam selector is configured to select one of the directional audio signals based on voice activity, echo detection, and signal energy.

An echo cancellation data synchronization control method is disclosed. The method includes: estimating a sound card delay value; initializing and accumulating a near-end audio buffer queue and a reference audio buffer queue until a difference obtained by subtracting a length of the near-end audio buffer queue from a length of the reference audio buffer queue is greater than or equal to an audio data length corresponding to the sound card delay value; extracting, in accordance with audio frames, audio data from the head of the reference audio buffer queue and the head of the near-end audio buffer queue for echo cancellation processing; obtaining a relative delay value generated by performing echo cancellation processing; and adjusting the sound card delay value according to the relative delay value.

An echo cancellation data synchronization control method is disclosed. The method includes: estimating a sound card delay value; initializing and accumulating a near-end audio buffer queue and a reference audio buffer queue until a difference obtained by subtracting a length of the near-end audio buffer queue from a length of the reference audio buffer queue is greater than or equal to an audio data length corresponding to the sound card delay value; extracting, in accordance with audio frames, audio data from the head of the reference audio buffer queue and the head of the near-end audio buffer queue for echo cancellation processing; obtaining a relative delay value generated by performing echo cancellation processing; and adjusting the sound card delay value according to the relative delay value.

From a mixed signal in which a first signal and a second signal are mixed, the second signal is removed at low processing cost and without delay. As a result, an estimated first signal which has low residue of the second signal and low distortion is obtained. An estimated first signal is generated by subtracting a pseudo second signal which is estimated to be mixed in a first mixed signal in which a first signal and a second signal are mixed from the first mixed signal. The pseudo second signal is obtained by a first adaptive filter using a second mixed signal in which the first signal and the second signal are mixed in a different proportion from the first mixed signal. A coefficient update amount of the first adaptive filter is made smaller as compared with a case when the estimated first signal is smaller than the first mixed signal, in case the estimated first signal is larger than the first mixed signal.

An audio echo suppressor includes a first receiver for receiving a first audio signal for rendering by a loud-speaker and a second receiver for receiving a microphone signal. A linear echo-cancellation filter generates a first compensation signal from the first audio signal and a compensator generates a residual signal by compensating the microphone signal for the first compensation signal. A first adapter determines filter parameters for the linear echo-cancellation filter. An estimator generates distortion measures where each distortion measure is indicative of the contribution to the residual signal in a frequency interval outside a first frequency interval. The residual signal results from rendering of signal components of the first audio signal that are within the first frequency interval. An echo reducer performs echo suppression based on the distortion measures.

Method and system for use with audible signals that analyzes the signals into time-frequency frames over first and second time periods. Estimates of the noise and/or reverberation are derived from the frames. Gains are derived from the estimates and raised to a power to create modified gains. The modified gains are applied to the frames in the appropriate time periods. Modified audible signals are output after being processed by the modified gains.

A packet voice transceiver adapted to reside at a first end of a communication network and to send and receive voice packets to and from a second packet voice transceiver residing at a second end of the communication network. The transceiver includes a voice encoder, a packet transmitter, a packet receiver, a voice decoder and a far-end echo canceller. The voice encoder receives an ingress audio signal and encodes the audio signal to produce an ingress packet voice signal. The packet transmitter transmits the ingress packet voice signal over the network to the second packet voice transceiver. The packet receiver receives an egress packet voice signal transmitted over the network by the second packet voice transceiver. The voice decoder decodes the egress packet voice signal to produce an egress audio signal. The far-end echo canceller reduces echo that is present in the egress audio signal and that originated at the second end.

A sound processing method performs near-end side sound collection processing to collect a sound on a near-end side and generate a sound collection signal, near-end side filter processing to adjust the sound collection signal using transfer characteristics of a far-end side, and near-end side sound emission processing to emit the sound collection signal that has been adjusted, from a speaker on the near-end side.

Communications equipment for use by a call center to enable communications between the call center and one or more user devices is provided, with the communications equipment comprising audio processing circuitry for processing audio signals received from the user devices over a communications network, wherein the processing circuitry is configured to perform acoustic echo cancellation and/or noise suppression and/or dereverberation processing on the received audio signals to cancel acoustic echoes and/or suppress noise and/or suppress reverberation that is present in the audio signals received from the user device.

A headphone can include plurality of exterior microphones, that generates corresponding exterior microphone signals, an accelerometer that generates an accelerometer signal; and an interior microphone, not directly exposed to the environment, that generates an interior microphone signal. A processor of the headphone can be configured to generate an audio signal containing voice of a user, based on a) the accelerometer signal, b) the interior microphone signal, and c) the plurality of exterior microphone signals.