A noise suppression device transforms observation signals to spectral components of multiple channels, calculates an arrival time difference, calculates weight coefficients based on the arrival time difference, estimates whether each of the spectral components of the plurality of frames is a spectral component of target sound or not, estimates a weighted S/N ratio of each of the spectral components of the plurality of frames based on the result of the estimation and the weight coefficients, calculates gains of the spectral components of the plurality of frames by using the weighted S/N ratios, outputs spectral components of an output signal by suppressing spectral components of observation signals of sounds other than the target sound in the spectral components of the plurality of frames by using the gains, and transforms the spectral components of the output signal to an output signal in a time domain.

A hearing aid includes a feedback control system for handling external feedback from an output transducer to an input transducer. The feedback control system includes an open loop gain estimator for providing an instant open loop gain estimate; an adaptive filter configured to provide a current estimate of the feedback path transfer function; a feedback change estimator configured to provide an instant estimate of the feedback path transfer function in dependence of the forward path transfer function, the instant open loop gain estimate; and an adaptive filter controller for providing an update transfer function estimate for the adaptive filter in dependence of the instant estimate of the feedback path transfer function. The hearing aid is configured to use the update transfer function estimate in the adaptive filter to update the current estimate of the feedback path transfer function. A method of detecting a sudden change in a feedback/echo path is further disclosed.

A sound source localization apparatus 2 includes a sound signal vector generation part 21 that generates a sound signal vector based on a plurality of electrical signals outputted from a plurality of microphones 11 that receive a sound generated by a sound source, a subspace identification part 22 that identifies a signal subspace corresponding to a signal component included in the sound signal vector and a noise subspace corresponding to a noise component included in the sound signal vector, a candidate identification part 23 that identifies one or more candidate vectors indicating a plurality of candidates of a direction of the sound source, and a direction identification part 24 that identifies, as the direction of the sound source, a direction, on the basis of an optimization objective function including a sum of squares of an inner product of the signal subspace and the noise subspace.

An audio generation method and system provided in this disclosure can dynamically select a frequency splicing point of an audio signal based on voice quality of a first audio signal and a second audio signal corresponding to each frequency in a frequency domain, divide the frequency domain into a first frequency interval and a second frequency interval, select audio signals of higher voice quality that correspond to each frequency interval for splicing, and obtain a target audio signal after fusion of the first audio signal and the second audio signal, so that voice quality of the target audio signal in each frequency interval in the frequency domain is the best, thereby improving voice quality of the target audio signal after fusion.

Some demonstrative embodiments include apparatuses, systems and/or methods of testing an acoustic device, an acoustic component, and/or a device or system including one or more acoustic devices. For example, an acoustic device tester may be configured to process input acoustic information of a tested acoustic device to determine a tested acoustic value distribution for the tested acoustic device in a plurality of frequency sub-bands; determine whether or not the tested acoustic device meets a predefined testing criterion based on the tested acoustic value distribution and a reference profile defining a plurality of reference values corresponding to the plurality of frequency sub-bands, respectively; and generate an output to indicate whether or not the tested acoustic device meets the predefined testing criterion.

The present disclosure relates to a device for processing an audio signal. The device may include a first acoustic-electric transducer and a second acoustic-electric transducer. The first acoustic-electric transducer may have a first frequency response, and may be configured to detect the audio signal and generate a first sub-band signal according to the detected audio signal. The second acoustic-electric transducer may have a second frequency response, the second frequency response being different from the first frequency response. The second acoustic-electric transducer may be configured to detect the audio signal and generate a second sub-band signal according to the detected audio signal.

A sound inspection system that can reduce power consumption is provided. A sound inspection system 1 that determines a state based on a sound of an inspection target object 2 includes a sound sensor device 10 that collects the sound of the inspection target object 2, analyzes the collected sound, and transmits an analysis result, and a sound data determination device 30 that determines a state of the inspection target object based on the analysis result from the sound sensor device. The sound sensor device transmits, as the analysis result, intensities of the collected sound for each predetermined frequency set in advance.

An electronic apparatus is provided. The electronic apparatus includes a communication interface; a display; an internal speaker; and a processor configured to control the internal speaker to generate a first audio signal and control the communication interface to transmit a signal to control an external audio device to generate a second audio signal that is synchronized with the first audio signal. The processor is configured to: obtain a first processing time regarding the first audio signal and a second processing time regarding the second audio signal; and control the internal speaker and the communication interface based on the first processing time and the second processing time so that a difference between a first output timing of the first audio signal and a second output timing of the second audio signal is less than a predetermined value.

A method for processing music audio data, including providing input audio data representing a first piece of music comprising a mixture of musical timbres. The method also includes decomposing the input audio data to generate at least first-timbre decomposed data representing a first timbre selected from the musical timbres of the first piece of music, and second-timbre decomposed data representing a second timbre selected from the musical timbres of the first piece of music. The method also includes applying a transport control to obtain transport controlled first-timbre decomposed data. The method also includes recombining audio data obtained from the transport controlled first-timbre decomposed data with audio data obtained from the second-timbre decomposed data to obtain recombined audio data.

A device for performing a hearing test on a test subject having a processing resource configured to: provide audio stimulus signals for an audio output device thereby to provide an audio stimulus to the test subject, wherein the audio stimulus signals are processed such that the provided audio stimulus has a simulated source location; obtain directional response data representative of a response of the test subject to the audio stimulus; process the obtained directional response data as part of determining the hearing ability of the test subject.