Abstract: The present disclosure is related to a method and system for a room calibration in a speaker system using an internal microphone. The method comprises calculating an impulse response of a sound signal received at an internal microphone from at least one speaker, and performing the room calibration based on the calculated impulse response.

Abstract: An audio playback device comprises a microphone, a speaker, and a processor. The processor is arranged to output by the speaker first audio content and receive by the microphone an indication of the first audio content. A first acoustic response of a room in which the audio playback device is located is determined based on the received indication of first audio content. A mapping is applied to the first acoustic response to determine a second acoustic response. The second acoustic response is indicative of an approximated acoustic response of the room at a spatial location different from a spatial location of the microphone. The second audio content output by the speaker is adjusted based on the second response.

Abstract: An example playback device is configured to receive a first stream of audio comprising source audio content to be played back by the playback device and record, via one or more microphones of the playback device, an audio signal output by the playback device based on the playback device playing the source audio content. The playback device is also configured to determine a transfer function between a frequency-domain representation of the first stream of audio and a frequency-domain representation of the recorded audio signal, and then determine an estimated frequency response of the playback device based on a difference between (i) the transfer function and (ii) a self-response of the playback device, where the self-response of the playback device is stored in a memory of the playback device. Based on the estimated frequency response, the playback device is configured to determine an acoustic calibration adjustment and implement the acoustic calibration adjustment.

Abstract: The disclosure relates to a soundbar and a method for automatic surround pairing and calibration of a surround sound system. The soundbar includes two built-in microphones on the left and right respectively, which can be used for determining relative positions of left and right surround speakers. When the relative positions of the left and right speakers are not correct, configurations of left and right surround channels can be automatically swapped with each other without manually swapping physical positions of the surround speakers by a user. In addition, latencies including a latency of each channel of a main system may also be calibrated, and magnitude compensation may be achieved by calculating a filter compensation coefficient of each line and merging it into an original filter. The automatic surround pairing and calibration of the surround sound system may be one-click completed automatically by the user by pressing a start button.

Abstract: A low complexity howling suppression system and method for portable karaoke system are provided. In the howling suppression, at least one infinite impulse response (IIR) filters are introduced for estimating the acoustic feedback picked up by the microphone from the real environment, and thereby to cancel out the acoustic feedback from the microphone input signal.

Abstract: In at least one embodiment, a system for providing an adaptive loudspeaker assembly is provided. A loudspeaker array transmits an audio output signal in an omnidirectional sound mode in a room having a plurality of walls. A microphone array is coupled to the loudspeaker array to capture the audio output signal in the room. At least one controller is programmed to receive the captured audio output signal and to determine that at least one first wall of the plurality of walls is closest to the loudspeaker array based on the captured audio output signal. The at least one controller is further programmed to change a sound mode of the loudspeaker array from transmitting the audio output signal in the omnidirectional mode into a beamforming sound mode to transmit the audio output signal away from the at least one first wall of the plurality walls.

Abstract: A vocal removal method and a system thereof are provided. In the vocal removal method, a voice separation model is generated and trained to process a real-time input music to separate the voice and the accompaniment. The vocal removal method further comprises the steps of feature extraction and reconstruction to obtain the voice minimized music.

Abstract: An example playback device is configured to receive a first stream of audio comprising source audio content to be played back by the playback device and record, via one or more microphones of the playback device, an audio signal output by the playback device based on the playback device playing the source audio content. The playback device is also configured to determine a transfer function between a frequency-domain representation of the first stream of audio and a frequency-domain representation of the recorded audio signal, and then determine an estimated frequency response of the playback device based on a difference between (i) the transfer function and (ii) a self-response of the playback device, where the self-response of the playback device is stored in a memory of the playback device. Based on the estimated frequency response, the playback device is configured to determine an acoustic calibration adjustment and implement the acoustic calibration adjustment.

Abstract: In one aspect, a first playback device is configured to (i) receive a set of voice signals, (ii) process the set of voice signals using a first set of audio processing algorithms, (iii) identify, from the set of voice signals, at least two voice signals that are to be further processed, (iv) determine that the first playback device does not have a threshold amount of computational power available, (v) receive an indication of an available amount of computational power of a second playback device, (vi) send the at least two voice signals to the second playback device, (vii) cause the second playback device to process the at least two voice signals using a second set of audio processing algorithms, (viii) receive, from the second playback device, the processed at least two voice signals, and (ix) combine the processed at least two voice signals into a combined voice signal.

Abstract: A method includes performing an authentication for a user based on a headphone transfer function (HPTF) of the user when the user wears the headphone. The method includes detecting whether a frequency response deviation exists between the HPTF of the user and a tuned HPTF. The method includes dynamically compensating for the HPTF of the user based on the detected frequency response deviation.

Abstract: In at least one embodiment, a system for providing an adaptive loudspeaker assembly is provided. A loudspeaker array transmits an audio output signal in an omnidirectional sound mode in a room having a plurality of walls. A microphone array is coupled to the loudspeaker array to capture the audio output signal in the room. At least one controller is programmed to receive the captured audio output signal and to determine that at least one first wall of the plurality of walls is closest to the loudspeaker array based on the captured audio output signal. The at least one controller is further programmed to change a sound mode of the loudspeaker array from transmitting the audio output signal in the omnidirectional mode into a beamforming sound mode to transmit the audio output signal away from the at least one first wall of the plurality walls.

Abstract: The present disclosure provides a method and system for voice enhancement. The method and system of the present disclosure may simultaneously perform signal processing of two paths on an input signal. The first path signal processing includes receiving an audio source input and performing dynamic loudness balancing on the audio source input based on a first gain control parameter. The second path signal processing includes: performing voice detection on the audio source input and calculating a detection confidence; and calculating a second gain control parameter based on the detection confidence. The first path signal processing and the second path signal processing may be synchronous or asynchronous. The method of the present disclosure also includes updating the first gain control parameter with the second gain control parameter calculated by a second processing path and performing the first path signal processing based on the updated first gain control parameter.

Abstract: The present disclosure provides an adaptive equalization method and system for an audio system. The method includes: detecting an input signal and estimating signal energy of the input signal; determining an audio signal status based on the estimated signal energy; performing dynamic gain calculation based on the estimated signal energy and the audio signal status; and using the calculated gain to adjust equalization parameters, thereby performing dynamic equalization processing on the input signal.

Abstract: In at least one embodiment, an apparatus for providing a virtual sound effect in a listening environment is provided. The apparatus includes at least one controller and an audio playback device. The audio playback device includes the at least one controller that is programmed to receive an audio input signal from an audio input source and to apply a head related transfer function (HRTF) to the audio input signal. The at least one controller is further programmed to apply crosstalk cancellation to the audio input signal and to generate an audio output signal after applying the HRTF and the crosstalk cancellation to the audio input signal for playback by at least one loudspeaker that is invisible to a listener in the listening environment.

Abstract: In one aspect, a first playback device is configured to (i) receive a set of voice signals, (ii) process the set of voice signals using a first set of audio processing algorithms, (iii) identify, from the set of voice signals, at least two voice signals that are to be further processed, (iv) determine that the first playback device does not have a threshold amount of computational power available, (v) receive an indication of an available amount of computational power of a second playback device, (vi) send the at least two voice signals to the second playback device, (vii) cause the second playback device to process the at least two voice signals using a second set of audio processing algorithms, (viii) receive, from the second playback device, the processed at least two voice signals, and (ix) combine the processed at least two voice signals into a combined voice signal.

Abstract: The disclosure relates to a soundbar and a method for automatic surround pairing and calibration of a surround sound system. The soundbar includes two built-in microphones on the left and right respectively, which can be used for determining relative positions of left and right surround speakers. When the relative positions of the left and right speakers are not correct, configurations of left and right surround channels can be automatically swapped with each other without manually swapping physical positions of the surround speakers by a user. In addition, latencies including a latency of each channel of a main system may also be calibrated, and magnitude compensation may be achieved by calculating a filter compensation coefficient of each line and merging it into an original filter. The automatic surround pairing and calibration of the surround sound system may be one-click completed automatically by the user by pressing a start button.

Abstract: A method of room calibration comprises measuring a plurality of impulse responses at a plurality of measurement points in a room for each speaker of a plurality of speakers. The method also comprises determining a plurality of transfer functions at the plurality of measurement points for each speaker based on the plurality of impulse responses. Furthermore, the method also comprises weighting and summing the transfer functions to obtain a weighted and summed sound curve for each speaker.

Abstract: The disclosure provides a method and a system for head-related transfer function (HRTF) adaptation. The method includes performing a system identification. The system identification includes a pinna identification and a shadowing identification.

Abstract: An audio playback device comprises a microphone, a speaker, and a processor. The processor is arranged to output by the speaker first audio content and receive by the microphone an indication of the first audio content. A first acoustic response of a room in which the audio playback device is located is determined based on the received indication of first audio content. A mapping is applied to the first acoustic response to determine a second acoustic response. The second acoustic response is indicative of an approximated acoustic response of the room at a spatial location different from a spatial location of the microphone. The second audio content output by the speaker is adjusted based on the second response.

Abstract: The present disclosure is related to a method and system for a room calibration in a speaker system using an internal microphone. The method comprises calculating an impulse response of a sound signal received at an internal microphone from at least one speaker, and performing the room calibration based on the calculated impulse response.