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: 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: 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: The disclosure relates to a method and system of virtualized spatial audio. The method may use a motion sensor to track a listener's movement, obtain location information associated with the listener's movement, and produce virtual sound adaptively based on the location information associated with the listener's movement. The location information may include distance information and direction information regarding the listener relative to the motion sensor.

Abstract: The disclosure describes a method for a multi-channel speaker system including N speakers. The method may comprise obtaining N! permutations of channel sequence for the N speakers; determining, for each permutation, a voting score that represents the matching degree between the channel sequence indicated in a permutation and a correct channel assignment sequence of the N speakers; selecting the permutation with the highest voting score; and assigning input source channels to the N speakers in the order of the channel sequence indicated in the selected permutation.

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 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: A method includes receiving, as input, reference audio data representing a reference audio signal captured by an audio input device. The method also includes receiving, as input, from a beamformer, spatially-filtered audio data representing an output of the beamformer, the beamformer configured to spatially filter, based on additional audio data captured by one or more additional audio input devices, the reference audio data to attenuate one or more interfering signals in the spatially-filtered audio data. The method processes, using a trained guided speech-enhancement network, the reference audio data and the spatially-filtered audio data to generate, as output, enhanced audio data, the guided speech-enhancement network processing the reference audio data and the spatially-filtered audio data to further attenuate, in the enhanced audio data, the one or more interfering signals attenuated by the beamformer.

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: Yes A keyword spotting method based on a neural network (NN) acoustic model is provided to tolerate for dynamically adding and deleting keywords by remapping new keywords as individual Acoustic Model Sequences. The method compares sequence matching in the phoneme spaces instead of directly in predetermined acoustic space. Therefore, the acoustic model cross comparison model is relaxed from global optimization to local minimum distance to each distribution.

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: 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.