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.

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 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.

Provided are an echo delay time estimation method and system thereof, wherein the echo delay time estimation method is executed by the echo delay time estimation system with the following steps: receiving a testing signal and a received signal and executing a time to frequency analysis to generate a testing signal spectrogram and a received signal spectrogram; respectively executing a characteristic signal dynamic detection calculation for the testing signal spectrogram and the received signal spectrogram to generate a testing signal characteristic dynamic vector and a received signal characteristic dynamic vector; executing a cross-correlated vector estimation for the testing signal characteristic dynamic vector and the received signal characteristic dynamic vector to generate a cross-correlated vector; and calculating an echo delay time according to the cross-correlated vector. The echo delay time estimation method is able to simplify calculations for estimating an echo delay time, alleviating some computational complexity.

An echo cancellation system that synchronizes output audio data with input audio data in a heterogeneous system. The system may increment a counter as outgoing audio frames are sent to a digital-to-analog converter in a speaker. As incoming audio frames are received by an analog-to-digital converter in a microphone, the system may copy contents of the counter into the incoming audio frames. Based on the contents of the counter, the incoming audio frames may be associated with corresponding outgoing audio frames. After synchronizing the incoming audio frames and the outgoing audio frames, the system may perform Acoustic Echo Cancellation by removing the outgoing audio frames from the incoming audio frames.

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.

A signal processing device includes an echo estimation device, a captured signal buffer device and a delay estimation device. The echo estimation device generates an echo estimation signal according to a reference signal and a set of reflection path simulation coefficients and compensates the echo estimation signal according to a first delay to generate a compensated echo estimation signal. The captured signal buffer device buffers a captured signal captured by microphone device and outputs the captured signal according to a second delay to generate a compensated captured signal. The delay estimation device estimates an amount of delay adjustment according to the compensated echo estimation signal and the compensated captured signal and updates the first delay or the second delay according to the amount of delay adjustment. A difference between an upper bound and a lower bound of the first delay is smaller than or equal to 1.