A method for improving speech signal intelligibility is performed at a device. A speech signal is obtained. A correspondence between the speech signal and a respective user group among different user groups having distinct voice characteristics is identified. Pre-encoding signal augmentation is performed on the speech signal with a respective pre-augmentation filtering coefficient that corresponds to the respective user group to obtain a group-specific pre-augmented speech signal. The device encodes the pre-augmented speech signal for subsequent transmission through the voice communication channel. An encoded version of the pre-augmented speech signal has reduced loss of signal quality as compared to an encoded version of the speech signal that is obtained without the pre-encoding signal augmentation.

A method for improving a listening experience of a user wearing a hearing device of a hearing system includes providing an audio sample based on a generated audio signal; determining, whether the audio sample comprises audio content matching a predetermined sound pattern indicative of an original version of the audio content; comparing the audio sample with the original version of the audio content, if the audio sample is determined to comprise the audio content matching the predetermined sound pattern; determining at least one acoustic property of current surroundings of the user depending on the comparison; determining at least one parameter for modifying the audio signal depending on the determined acoustic property; modifying the audio signal generated by the sound input component by the sound processing module, in accordance with the determined parameter; and outputting the modified audio signal to the user.

A method for improving a listening experience of a user wearing a hearing device of a hearing system includes providing an audio sample based on a generated audio signal; determining, whether the audio sample comprises audio content matching a predetermined sound pattern indicative of an original version of the audio content; comparing the audio sample with the original version of the audio content, if the audio sample is determined to comprise the audio content matching the predetermined sound pattern; determining at least one acoustic property of current surroundings of the user depending on the comparison; determining at least one parameter for modifying the audio signal depending on the determined acoustic property; modifying the audio signal generated by the sound input component by the sound processing module, in accordance with the determined parameter; and outputting the modified audio signal to the user.

Acoustic echo cancellation systems and methods are provided that can cancel and suppress acoustic echo from the output of a mixer that has mixed audio signals from a plurality of acoustic sources, such as microphones. The microphones may have captured speech and sound from a remote location or far end, such as in a conferencing environment. The acoustic echo cancellation may generate an echo-cancelled mixed audio signal based on a mixed audio signal from a mixer, information gathered from the audio signal from each of the plurality of acoustic sources, and a remote audio signal. The systems and methods may be computationally efficient and resource-friendly.

Acoustic echo cancellation systems and methods are provided that can cancel and suppress acoustic echo from the output of a mixer that has mixed audio signals from a plurality of acoustic sources, such as microphones. The microphones may have captured speech and sound from a remote location or far end, such as in a conferencing environment. The acoustic echo cancellation may generate an echo-cancelled mixed audio signal based on a mixed audio signal from a mixer, information gathered from the audio signal from each of the plurality of acoustic sources, and a remote audio signal. The systems and methods may be computationally efficient and resource-friendly.

One example includes a digital twin of a microphone array. The digital twin acts as a digital copy of a physical microphone array. The digital array allows the microphone array to be analyzed, simulated and optimized. Further, the microphone array can be optimized for performing sound quality operations such as noise suppression and speech intelligibility.

A method for speech equalization, comprising the steps of receiving an input audio signal, processing said input audio signal in dependence on frequency and to providing an equalized electric audio signal according to an equalization function, wherein said equalization function comprises at least an actuator part configured to dynamically applying a compensation filter to the received input signal and dynamically applying a transparent filter to the received input signal, and further transmitting an output signal perceivable by a user as sound representative of said electric acoustic input signal or a processed version thereof.

A method and an apparatus for audio signal processing evaluation are provided. The audio signal processing is performed on a synthesized audio signal to generate a processed audio signal. The synthesized audio signal is generated by adding a secondary signal into a master signal. The master signal is merely a speech signal. The signal processing is related to removing the secondary signal from the synthesized audio signal. The sound characteristics of the processed audio signal and the master signal are obtained, respectively. The sound characteristics include text content, and the text content is generated by performing speech-to-text on the processed audio signal and the master signal. The audio signal processing is evaluated according to the compared result between the sound characteristics of the processed audio signal and the master signal. The compared result includes the correctness of the text content of the processed audio signal relative to the master signal.

A method and an apparatus for audio signal processing evaluation are provided. The audio signal processing is performed on a synthesized audio signal to generate a processed audio signal. The synthesized audio signal is generated by adding a secondary signal into a master signal. The master signal is merely a speech signal. The signal processing is related to removing the secondary signal from the synthesized audio signal. The sound characteristics of the processed audio signal and the master signal are obtained, respectively. The sound characteristics include text content, and the text content is generated by performing speech-to-text on the processed audio signal and the master signal. The audio signal processing is evaluated according to the compared result between the sound characteristics of the processed audio signal and the master signal. The compared result includes the correctness of the text content of the processed audio signal relative to the master signal.

A vehicle includes a plurality of microphones to obtain speech from a person outside the vehicle as an input signal and a sensor system to determine a location and orientation of the person relative to the vehicle. The vehicle also includes a controller to determine characteristics of the input signal and to determine whether to perform speech enhancement on the input signal based on one or more of the characteristics and the location and orientation of the person.