Techniques for adaptively providing acoustic echo cancellation (AEC) for a stereo audio signal associated with at least one microphone are discussed herein. Some embodiments may include determining, based at least in part on detecting a reference signal associated with a channel sample portion of the stereo audio signal, a panning state of the stereo audio signal. A hard-panned-configured AEC processing filter or a soft-panned-configured AEC processing filter is applied to the stereo audio signal to generate a filtered audio signal output based on the panning state.

An embodiment for cancelling echo in online conference systems is provided. According to some embodiments of the present disclosure, the computer-implemented method comprises, in response to an update of devices of participants in an online conference, dividing, by one or more processors, the devices in an online conference into a plurality of groups, wherein the devices located in a same physical location are divided into a same group. The method also comprises, in response to an update of the devices in an online conference, selecting at least one speaker of the devices in each of the plurality of groups as a representative speaker for each of the plurality of groups. The method further comprises forwarding audio data received from microphones of the devices in one of the plurality of groups to the respective representative speaker of other groups of the plurality of groups.

A first mobile communication device that includes a first microphone, a first speaker, and a first delay unit. The first microphone is configured to (i) receive, during a conference call, a first user first microphone signal from a first user, and (ii) output a first microphone digital signal to the first delay unit. The first user first microphone signal represents audio content outputted by the first user. The first delay unit is configured to delay, by a delay period, the first microphone digital signal to provide a delayed first user first device digital signal. The first mobile communication device is configured to output, to a mixer, the delayed first user first device digital signal. The delay period is determined based on measurements executed by at least one mobile communication device out of the first mobile communication device, a second mobile communication device and a third mobile communication device.

Some disclosed teleconferencing methods may involve detecting a howl state during a teleconference. The teleconference may involve two or more teleconference client locations and a teleconference server. The teleconference server may be configured for providing full-duplex audio connectivity between the teleconference client locations. The howl state may be a state of acoustic feedback involving two or more teleconference devices in a teleconference client location. Detecting the howl state may involve an analysis of both spectral and temporal characteristics of teleconference audio data. Some disclosed teleconferencing methods may involve determining which client location is causing the howl state. Some such methods may involve mitigating the howl state and/or sending a howl state detection message.

A variable update step size is determined in proportion to a magnitude ratio or magnitude difference between a first residual signal and a second residual signal. The first residual signal is obtained by using adaptive filter coefficient sequence, where the adaptive filter coefficient sequence has been obtained in previous operations of the adaptive equalizer. The second residual signal is obtained by using a prior update adaptive filter coefficient sequence, where the prior update adaptive filter coefficient sequence is obtained by performing a coefficient update with an arbitrary prior update step size on the adaptive filter coefficient sequence having been obtained in previous operations of the adaptive equalizer.

The present invention relates to a sound effect control method and apparatus, where the method includes: when a hands-free call is performed for a mobile terminal, detecting whether a hands-free call channel of the mobile terminal is shielded; and adjusting a configuration of the hands-free call channel of the mobile terminal and/or outputting an alarm signal to inform a user that the hands-free call channel is shielded, when it is detected that the hands-free call channel of the mobile terminal is shielded. According to the method, when a hands-free call is performed for a mobile terminal, whether a hands-free call channel of the mobile terminal is shielded is detected; when it is shielded, a configuration of the hands-free call channel of the mobile terminal is adjusted and/or an alarm signal is output, which can effectively improve quality of the hands-free call of the mobile terminal.

Echo reduction. At least one example embodiment is a method including producing, by a loudspeaker, acoustic waves based on a far-microphone signal; receiving, at a local microphone, an echo based on the acoustic waves, and receiving acoustic waves generated locally, the receiving creates a local-microphone signal; producing an estimated-echo signal based on the far-microphone signal and a current step-size parameter; summing the local-microphone signal and the estimated echo signal to produce a resultant signal having reduced echo in relation to the local-microphone signal; and controlling the current step-size parameter. The controlling current step size may include: calculating a convergence value based on a cross-correlation of the local-microphone signal and the resultant signal; and updating the current step-size parameter based on the convergence value.

In example implementations, an apparatus is provided. The apparatus includes an adaptive filter and a double talk detector in communication with the adaptive filter. The adaptive filter is to calculate a transfer function with coefficients for a particular time that is applied to an output signal of a microphone to cancel echoes caused by a reference signal in the output signal of the microphone. The double talk detector is to determine a peak of the coefficients, detect double talk based on a location of the peak of the coefficients, and transmit a pause signal to the adaptive filter in response to detection of the double talk, wherein the pause signal is to pause a calculation of updates to the coefficients by the adaptive filter.

In example implementations, an apparatus is provided. The apparatus includes an adaptive filter and a double talk detector in communication with the adaptive filter. The adaptive filter is to calculate a transfer function with coefficients for a particular time that is applied to an output signal of a microphone to cancel echoes caused by a reference signal in the output signal of the microphone. The double talk detector is to determine a peak of the coefficients, detect double talk based on a location of the peak of the coefficients, and transmit a pause signal to the adaptive filter in response to detection of the double talk, wherein the pause signal is to pause a calculation of updates to the coefficients by the adaptive filter.

In an audio processing system (300), a filtering section (350, 400): receives subband signals (410, 420, 430) corresponding to audio content of a reference signal (301) in respective frequency subbands; receives subband signals (411, 421, 431) corresponding to audio content of a response signal (304) in the respective subbands; and forms filtered inband references (412, 422, 432) by applying respective filters (413, 423, 433) to the subband signals of the reference signal. For a frequency subband: filtered crossband references (424, 425) are formed by multiplying, by scalar factors (426, 427), filtered inband references of other subbands; a composite filtered reference (428) is formed by summing the filtered inband reference of the subband (422) and the filtered crossband references; a residual signal (429) is computed as a difference between the composite filtered reference and the subband signal of the response signal corresponding to the subband; and the scalar factors and the filter applied to the subband signal of the reference signal corresponding to the subband are adjusted based on the residual signal.