Abstract: A method and a control circuit for performing synchronization on playback of multiple electronic devices are provided, where the multiple electronic devices include a master device and a slave device. The method includes: utilizing the master device to receive audio data from a far-end device to be master audio data, and transmitting the master audio data to the slave device to be slave audio data, where the slave device generates ultrasound data according to the slave audio data, to make the slave device play the slave audio data and the ultrasound data; utilizing a calibration circuit to estimate a delay value between the ultrasound data received by a microphone of the master device and the master audio data; and utilizing a delay circuit to control a delay of the master audio data according to the delay value, to generate master output audio data for being played by the master device.

Abstract: A voice capturing method includes following operations: storing, by a buffer, voice data from a plurality of microphones; determining, by a processor, whether a target speaker exists and whether a direction of the target speaker changes according to the voice data and target speaker information; inserting a voice segment corresponding to a previous tracking direction into a current position in the voice data to generate fusion voice data when the target speaker exists and the direction of the target speaker changes from the previous tracking direction to a current tracking direction; performing, by the processor, a voice enhancement process on the fusion voice data according to the current tracking direction to generate enhanced voice data; performing, by the processor, a voice shortening process on the enhanced voice data to generate voice output data; and playing, by a playing circuit, the voice output data.

Abstract: An audio playback method includes the following steps: receiving an ultrasound signal via a microphone; processing the ultrasound signal to obtain a characteristic value of the ultrasound signal; obtaining time lag information based on the characteristic value; and controlling audio delay according to the time lag information.

Abstract: An audio playback method includes the following steps: receiving an ultrasound signal via a microphone; processing the ultrasound signal to obtain a characteristic value of the ultrasound signal; obtaining time lag information based on the characteristic value; and controlling audio delay according to the time lag information.

Abstract: A voice capturing method includes following operations: storing, by a buffer, voice data from a plurality of microphones; determining, by a processor, whether a target speaker exists and whether a direction of the target speaker changes according to the voice data and target speaker information; inserting a voice segment corresponding to a previous tracking direction into a current position in the voice data to generate fusion voice data when the target speaker exists and the direction of the target speaker changes from the previous tracking direction to a current tracking direction; performing, by the processor, a voice enhancement process on the fusion voice data according to the current tracking direction to generate enhanced voice data; performing, by the processor, a voice shortening process on the enhanced voice data to generate voice output data; and playing, by a playing circuit, the voice output data.

Abstract: A device for sound localization includes a spatial feature generator, a voice detector, an angle selector, and an angle retriever. The spatial feature generator generates M spatial feature signals according to signals of N microphones of a microphone array. The voice detector generates at least one voice detection signal according to at least one of the signals of the N microphones. The angle selector outputs a candidate angle signal according to the M spatial feature signals to indicate a candidate direction of sound. The angle retriever generates a sound detection result according to the M spatial feature signals to indicate whether any sound source exists, and then outputs an estimated angle signal indicative of a direction of sound according to the sound detection result, the at least one voice detection signal, and the candidate angle signal.

Abstract: Disclosed is a device for sound localization. The device can determine a direction of sound adequately, and includes a spatial feature generator, a voice detector, an angle selector, and an angle retriever. The spatial feature generator generates M spatial feature signals according to signals of N microphones of a microphone array. The voice detector generates at least one voice detection signal according to at least one of the signals of the N microphones. The angle selector outputs a candidate angle signal according to the M spatial feature signals to indicate a candidate direction of sound. The angle retriever generates a sound detection result according to the M spatial feature signals to indicate whether any sound source exists, and then outputs an estimated angle signal indicative of a direction of sound according to the sound detection result, the at least one voice detection signal, and the candidate angle signal.

Abstract: A method for calculating excursion of a diaphragm of a speaker, which includes: calculating a DC resistance based on a feedback signal of the speaker; calculating an impedance based on the feedback signal and the DC resistance; calculating a transfer function between a signal and a velocity of the diaphragm of the speaker based on the DC resistance, the impedance, and a force factor; and calculating the excursion of the diaphragm of the speaker based on an input signal and the transfer function.

Abstract: A method of audio source separation includes steps of applying a demixing matrix on a plurality of received signals to generate a plurality of separated results; performing a recognition operation on the plurality of separated results to generate a plurality of recognition scores; generating a constraint according to the plurality of recognition scores; and adjusting the demixing matrix according to the constraint; where the adjusted demixing matrix is applied to the plurality of received signals to generate a plurality of updated separated results from the plurality of received signals.

Abstract: A method of audio source separation includes steps of applying a demixing matrix on a plurality of received signals to generate a plurality of separated results; performing a recognition operation on the plurality of separated results to generate a plurality of recognition scores; generating a constraint according to the plurality of recognition scores; and adjusting the demixing matrix according to the constraint; where the adjusted demixing matrix is applied to the plurality of received signals to generate a plurality of updated separated results from the plurality of received signals.

Abstract: A method for calculating excursion of a diaphragm of a loudspeaker is provided. The loudspeaker includes a diaphragm and is driven by a voltage signal. The method includes: a) low-pass filtering the voltage signal and a current signal inputted to the loudspeaker to generate a low-pass filtered voltage signal and a low-pass filtered current signal, respectively; b) calculating a direct-current (DC) resistance of the loudspeaker according to the low-pass filtered voltage signal and the low-pass filtered current signal; c) calculating a vibration velocity of the diaphragm according to the voltage signal, the current signal and the DC resistance; and d) calculating the excursion of the diaphragm according to the vibration velocity. Step (a) to step (d) involve real-number calculations.

Abstract: An apparatus for processing an audio input signal is provided and includes an audio processing circuit and an audio compressing circuit. The audio processing circuit receives the audio input signal, and enhances a first frequency part of the audio input signal to output a bass-enhancement signal. The audio compressing circuit is coupled to the audio processing circuit, and reduces a gain of a second frequency part of the bass-enhancement signal to output an audio output signal.

Abstract: An apparatus for processing an audio input signal is provided and includes an audio processing circuit and an audio compressing circuit. The audio processing circuit receives the audio input signal, and enhances a first frequency part of the audio input signal to output a bass-enhancement signal. The audio compressing circuit is coupled to the audio processing circuit, and reduces a gain of a second frequency part of the bass-enhancement signal to output an audio output signal.

Abstract: An audio coding and decoding device, which is used to realize the coding and decoding functions, audio coding and decoding device comprising: a first circuit, which is used to perform the first operation of said audio coding and decoding function; a general-purpose processor, which is used to perform the second operation of said audio coding and decoding function according to an instruction set and a software program; and a serial link, which is coupled between said first circuit and said general-purpose processor, and is used to serially transmitting the data between said first circuit and said general-purpose processor.