Abstract: A method and a system for intelligent dynamic speech enhancement for an audio source, comprising performing speech detection and intelligent enhancement gain control on a multi-channel audio source input to determine speech enhancement gain, and further comprising applying the speech enhancement gain in dynamic loudness balancing performed on the multi-channel audio source input, wherein the intelligent enhancement gain control comprises setting the speech enhancement gain based on a signal power strength ratio of a signal of a center channel to a sum of signals of other channels, and setting the speech enhancement gain based on a system volume level

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: A method for processing multi-channel audio signals, a system for processing multi-channel audio signals and a stereo apparatus comprising such a system. The multi-channel audio processing method comprises: receiving multi-channel audio signals from an external audio source, the multi-channel audio signals comprises a pair of surround channel signals and a pair of top channel signals; applying a crosstalk cancellation process in consideration of a head-related transfer function to the pair of top channel signals, so as to produce a pair of processed top channel signals, the head-related transfer function is configured to provide an elevation angle; mixing the pair of processed top channel signals, respectively, with the pair of surround channel signals, so as to produce a pair of mixed surround channel signals; providing the pair of mixed surround channel signals, respectively, to a pair of surround speakers.

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: 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: 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: 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: 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: A method and a system for instrument separating and reproducing for a mixture audio source is provided. The method and/or the system includes inputting selected music into an instrument separation model for extracting features therefrom, determining audio source signals of multiple channels for the separation of all instruments, each channel containing sound of one instrument, and transmitting the signals of the different channels to multiple speakers placed at designated positions for playing, which can reproduce or recreate an immersive sound field listening experience for users.

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: 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: A network data scheduling method applied to an edge node. The method includes: reading scheduling strategy information of a target domain name, the scheduling strategy information at least including a network scheduling proportion threshold; for a target connection that accesses the target domain name, setting a scheduling identification for the target connection, according to a current network scheduling proportion and the network scheduling proportion threshold under the target domain name; and wherein the scheduling identification is configured to indicate a network scheduling mode that is adopted for a data message transmitted through the target connection. The technical solution provided in the present disclosure can improve the quality of data transmission in networks.

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.