This disclosure falls into the field of audio coding, in particular it is related to the field of providing a framework for providing loudness consistency among differing audio output signals. In particular, the disclosure relates to methods, computer program products and apparatus for encoding and decoding of audio data bitstreams in order to attain a desired loudness level of an output audio signal.

Provided is a signal processing device including a main speech detection unit configured to detect, by using a neural network, whether or not a signal input to a sound collection device assigned to each of at least two speakers includes a main speech that is a voice of the corresponding speaker, and output frame information indicating presence or absence of the main speech.

Provided is a signal processing device including a main speech detection unit configured to detect, by using a neural network, whether or not a signal input to a sound collection device assigned to each of at least two speakers includes a main speech that is a voice of the corresponding speaker, and output frame information indicating presence or absence of the main speech.

A method and a system for processing an audio stream are described, wherein at least one database of classified voices and at least one database of classified background sounds are provided and a comparison between these classified voices and background sounds with the voices and the sounds extrapolated from a suitably re-processed audio stream is carried out in order to identify possible matches.

A method and a system for processing an audio stream are described, wherein at least one database of classified voices and at least one database of classified background sounds are provided and a comparison between these classified voices and background sounds with the voices and the sounds extrapolated from a suitably re-processed audio stream is carried out in order to identify possible matches.

Aspects of the subject technology relate to a device including a microphone, a filter and a processor. The filter receives an audio signal including ambient noise and a voice of a user of the device from the microphone. At least a portion of ambient noise is filtered from the audio signal. The processor determines a level of the ambient noise in the received audio signal and dynamically adjusts a gain applied to the filtered audio signal based on the level of the ambient noise.

Aspects of the subject technology relate to a device including a microphone, a filter and a processor. The filter receives an audio signal including ambient noise and a voice of a user of the device from the microphone. At least a portion of ambient noise is filtered from the audio signal. The processor determines a level of the ambient noise in the received audio signal and dynamically adjusts a gain applied to the filtered audio signal based on the level of the ambient noise.

This application relates to the field of signal processing technologies and headsets, and provides a speech signal processing method and apparatus, to provide a full-band low-noise speech signal. The method is applied to a headset including at least two speech collectors, where the at least two speech collectors include an ear canal speech collector and at least one external speech collector. The method includes: preprocessing a speech signal that is in a first frequency band and that is collected by the ear canal speech collector, to obtain a first speech signal; preprocessing a speech signal that is in a second frequency band and that is collected by the at least one external speech collector, to obtain an external speech signal, where frequency ranges of the first frequency band and the second frequency band are different; performing correlation processing on the first speech signal and the external speech signal to obtain a second speech signal; and outputting a target speech signal, where the target speech signal includes the first speech signal and the second speech signal.

This application provides a speech signal processing method and apparatus, and relates to the field of signal processing technologies and earphone, to monitor an ambient sound signal and improve a monitoring effect and user experience. The method is applied to an earphone, where the earphone includes at least one external speech collector. The method includes: preprocessing a speech signal collected by the at least one external speech collector, to obtain an external speech signal; extracting an ambient sound signal from the external speech signal; and performing audio mixing processing on a first speech signal and the ambient sound signal based on amplitudes and phases of the first speech signal and the ambient sound signal and a location of the at least one external speech collector, to obtain a target speech signal.

This application provides a speech signal processing method and apparatus, and relates to the field of signal processing technologies and earphone, to monitor an ambient sound signal and improve a monitoring effect and user experience. The method is applied to an earphone, where the earphone includes at least one external speech collector. The method includes: preprocessing a speech signal collected by the at least one external speech collector, to obtain an external speech signal; extracting an ambient sound signal from the external speech signal; and performing audio mixing processing on a first speech signal and the ambient sound signal based on amplitudes and phases of the first speech signal and the ambient sound signal and a location of the at least one external speech collector, to obtain a target speech signal.