This application relates to a howling suppression method and apparatus, a computer device, and a storage medium. The method includes obtaining a current audio signal corresponding to a current time period, and performing frequency domain transformation on the current audio signal; dividing the frequency domain audio signal and determining a target subband; obtaining a current howling detection result and a current voice detection result that correspond to the current audio signal, and determining a subband gain coefficient; obtaining a past subband gain corresponding to an audio signal within a past time period, and calculating a current subband gain corresponding to the current audio signal based on the subband gain coefficient and the past subband gain; and suppressing howling on the target subband based on the current subband gain, to obtain a first target audio signal corresponding to the current time period.

Echo cancellation for a two-way audio communication includes receiving, at an AEC system from microphone(s), an audio signal based on, at least in part, near-end signals and reproduced far-end signals. Loudspeaker(s) reproduced the far-end signals. The AEC system is operated, at least in part, with filter(s) so as to update estimates of coefficients of an acoustic channel from the loudspeaker(s) to the microphone(s). Control parameter(s) affecting an operation of the AEC system that is/are configurable and is/are set to value(s), from a range of values, is/are determined, based on estimating an accuracy of the estimates of the coefficients of the acoustic channel and a characteristic of the near-end signals. The AEC system controls the filter(s) with different values of the control parameter(s) at different times.

An AEC system can implement different updating techniques for echo cancellation. A Block Maximum Likelihood updating technique updates filter coefficients typically at integer multiples of a block size. A Proportional-Incremental Maximum Likelihood updating technique applies proportional weighting to the updates of filter coefficients. A Multi-source Incremental Maximum Likelihood updating technique uses separate control parameters associated with each of multiple loudspeakers. Combinations of these may be used, and one or more of these can be selected and implemented in real-time.

A sound emission and collection device includes a speaker, a filter processing a sound emission signal, microphones, echo cancellers cancelling regression sound signals of the sound emitted by the speaker from the sound collection signals of the corresponding microphones, a first integration section integrating adaptive filter coefficients taken out from the plurality of echo cancellers, a reverberation time estimation section estimating the reverberation time for each frequency band in the space in which the speaker and the plurality of microphones are present on the basis of the integrated adaptive filter coefficient, and an arithmetic operation section specifying a frequency band having a long reverberation time from the sound emission signal based on the estimated reverberation time, calculating a filter coefficient for suppressing power of the specified frequency band, and setting the filter coefficient to the filter.

Techniques of performing acoustic echo cancellation involve providing a bi-magnitude filtering operation that performs a first filtering operation when a magnitude of an incoming audio signal to be output from a loudspeaker is less than a specified threshold and a second filtering operation when the magnitude of the incoming audio signal is greater than the threshold. The first filtering operation may take the form of a convolution between the incoming audio signal and a first impulse response function. The second filtering operation may take the form of a convolution between a nonlinear function of the incoming audio signal and a second impulse response function. For such a convolution, the bi-magnitude filtering operation involves providing, as the incoming audio signal, samples of the incoming audio signal over a specified window of time. The first and second impulse response functions may be determined from an input signal input into a microphone.

A three-dimensional sound generation system includes one or more processors of a computing device, including capability to receive sound tracks, each of the sound tracks comprising one or more sound sources, each of the one or more sound sources corresponding to one or more respective sound categories, receive or determine a first configuration in a three-dimensional space, the first configuration comprising a listener position and a computing device location relative to the listener position, determine a second configuration comprising a change to at least one of the listener location or the computing device location relative to the listener position, generate, using the one or more sound tracks and the second configuration, one or more channels of sound signals, and provide the one or more channels of sound signals to drive one or more sound generation devices to generate a three-dimensional sound field.

The present disclosure relates to a device having an output transducer and an input transducer, wherein the device is configured to be operated so as to reduce or cancel signal content that is output via the output transducer in the input transducer signal.

Methods and devices for aggregating hardware loopback streams of a plurality of display devices in communication with a computer device may include a plurality of hardware loopback streams with rendered audio data from the plurality of display devices in communication with the computer device. The methods and devices may include combining the rendered audio data from the plurality of hardware loopback streams into a loopback buffer to create aggregated loopback audio data. The methods and devices may include providing the loopback buffer with the aggregated loopback audio data to one or more applications executing on the computer device.

A method is disclosed, the method comprising obtaining at least one first information indicative of audio data gathered by at least one first microphone, and at least one second information indicative of audio data gathered by at least one second microphone; determining a differential information indicative of one or more differences between at least two pieces of information, wherein the differential information is determined based, at least in part, on the at least one first information and the at least one second information; and compensating of an impact onto the audio data, wherein audio data of the first information and/or the second information is compensated based, at least in part, on the determined differential information. Further, an apparatus, and a system are disclosed.

To obtain an appropriate evaluation value in an acoustic quality evaluation by a conversational test. An acoustic quality evaluation apparatus 3 evaluates the acoustic quality of a call performed between a near-end terminal 1 and a far-end terminal 2 via a voice communication network 4. An evaluation value presenting unit 31 displays, on a display unit 13, evaluation categories obtained by classifying each of a plurality of evaluation viewpoints into a predetermined number of levels. An input unit 14 transmits the evaluation category selected by the evaluator for each of the evaluation viewpoints, to an evaluation value determination unit 32. The evaluation value determination unit 32 determines the lowest evaluation value among evaluation values assigned to the evaluation category received from the input unit 14 as a subjective evaluation value for acoustic quality.