A sound processing apparatus includes a first calculator that calculates first power based on a first signal received by a first microphone that is among the first microphone and a second microphone; a second calculator that calculates second power based on a second signal received by the second microphone; a gain calculator that calculates a gain on the basis of the ratio of the first power to the second power; and a multiplier that processes the second signal using the gain calculated by the gain calculator.

Audio pickup systems and methods are provided to enhance an audio signal by removing noise components related to an acoustic environment. The systems and methods receive a primary signal and one or more reference signals from various microphones. Adaptive filtering and combining minimizes an energy content of a resulting output signal, e.g., to form a substantially null output when the system is in a static acoustic environment. When the system is a playback sound source, one or more echo cancellers may contribute to removing content from the output signal. A change in the acoustic environment, such as a new sound source, causes content in the output signal until the adaptive filtering adapts to the new environment. In some examples, a desired content such as a wake-up word is detected and adaptation is stopped.

A high-performance method evaluates a useful signal of an audio device, and in particular of an audio apparatus, for example for reducing interference. Accordingly, in the method at least two microphone signals are each obtained from a sound signal and a reference signal is obtained from the microphone signals, a portion of the microphone signals from a predetermined direction being blocked. The microphone signals are filtered by a filter such that an evaluation signal is obtained. To that end, a coherence value is determined from portions of the reference signal and a power density value is determined from the coherence value. The filter is parameterized on the basis of the power density value.

An original noisy signal of each of at least two microphones is acquired by acquiring, using the at least two microphones, an audio signal emitted by each sound source. For each frame in time domain, an estimated frequency-domain signal of each sound source is acquired according to the original noisy signal of each of the at least two microphones. A frequency collection containing a plurality of predetermined static frequencies and dynamic frequencies is determined in a predetermined frequency band range. A weighting coefficient of each frequency contained in the frequency collection is determined according to the estimated frequency-domain signal of the each frequency in the frequency collection. A separation matrix of the each frequency is determined according to the weighting coefficient. The audio signal emitted by each of the at least two sound sources is acquired based on the separation matrix and the original noisy signal.

A mobile communications device contains at least two microphones. One microphone is designated by a selector to provide a voice dominant signal and another microphone is designated to provide a noise or echo dominant signal, for a call or a recording. The selector communicates the designations to a switch that routes the selected microphone signals to the inputs of a processor for voice signal enhancement. The selected voice dominant signal is then enhanced by suppressing ambient noise or canceling echo therein, based on the selected noise or echo dominant signal. The designation of microphones may change at any instant during the call or recording depending on various factors, e.g. based on the quality of the microphone signals. Other embodiments are also described.

The disclosure provides a method and apparatus for processing a voice signal. The method for processing a voice signal includes: acquiring first voice signals using the at least two voice acquiring devices; determining sound source feature values of the first voice signals acquired by the respective at least two voice acquiring devices; determining a voice processing scheme corresponding to the sound source feature values of the first voice signals acquired by the at least two voice acquiring devices according to a preset first correspondence relationship including a correspondence relationship between a range of source feature values corresponding to the at least two voice acquiring devices and a voice processing scheme; and processing the first voice signals acquired by the at least two voice acquiring devices according to the determined voice processing scheme.

A personal audio device, such as a wireless telephone, includes an adaptive noise canceling (ANC) circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone is also provided proximate to the speaker to provide an error signal indicative of the effectiveness of the noise cancellation. A secondary path estimating adaptive filter is used to estimate the electro-acoustical path from the noise canceling circuit through the transducer so that source audio can be removed from the error signal. A level of the source audio with respect to the ambient audio is determined to determine whether the system may generate erroneous anti-noise and/or become unstable.

A system and method for generating a self-steering beamformer is provided. Embodiments may include receiving, at one or more microphones, a first audio signal and adapting one or more blocking filters based upon, at least in part, the first audio signal. Embodiments may also include generating, using the one or more blocking filters, one or more noise reference signals. Embodiments may further include providing the one or more noise reference signals to an adaptive interference canceller to reduce a beamformer output power level.

Some of the embodiments of the present disclosure provide a device comprising: a first channel configured to receive a signal, wherein the signal comprises (i) a target signal and (ii) a background signal; a second channel configured to receive the signal a time t after the first channel receives the signal; a delay control circuit configured to iteratively determine a fractional delay to maximize a correlation coefficient between the signal on the first channel and the signal on the second channel; and an adaptive fractional delay filter in the first channel configured to adaptively align, in the digital domain, the signal on the first channel with the signal on the second channel based, at least in part, on the fractional delay.

A dual omnidirectional microphone array noise suppression is described. Compared to conventional arrays and algorithms, which seek to reduce noise by nulling out noise sources, the array of an embodiment is used to form two distinct virtual directional microphones which are configured to have very similar noise responses and very dissimilar speech responses. The only null formed is one used to remove the speech of the user from V.sub.2. The two virtual microphones may be paired with an adaptive filter algorithm and VAD algorithm. to significantly reduce the noise without distorting the speech, significantly improving the SNR of the desired speech over conventional noise suppression systems.