Dynamic adjustment of audio characteristics for enhancing musical sound during a networked conference is disclosed. In an embodiment, a method is provided for sound enhancement performed by a device coupled to a network. The method includes receiving an audio signal to be transmitted over the network, detecting when musical content is present in the audio signal, processing the audio signal to enhance voice characteristics to generate an enhanced audio signal when the musical content is not detected, processing the audio signal to enhance music characteristic to generate the enhanced audio signal when the musical content is detected, and transmitting the enhanced audio signal over the network.

Audio distortion compensation methods to improve accuracy and efficiency of audio content identification are described. The method is also applicable to speech recognition. Methods to detect the interference from speakers and sources, and distortion to audio from environment and devices, are discussed. Additional methods to detect distortion to the content after performing search and correlation are illustrated. The causes of actual distortion at each client are measured and registered and learnt to generate rules for determining likely distortion and interference sources. The learnt rules are applied at the client, and likely distortions that are detected are compensated or heavily distorted sections are ignored at audio level or signature and feature level based on compute resources available. Further methods to subtract the likely distortions in the query at both audio level and after processing at signature and feature level are described.

Audio distortion compensation methods to improve accuracy and efficiency of audio content identification are described. The method is also applicable to speech recognition. Methods to detect the interference from speakers and sources, and distortion to audio from environment and devices, are discussed. Additional methods to detect distortion to the content after performing search and correlation are illustrated. The causes of actual distortion at each client are measured and registered and learnt to generate rules for determining likely distortion and interference sources. The learnt rules are applied at the client, and likely distortions that are detected are compensated or heavily distorted sections are ignored at audio level or signature and feature level based on compute resources available. Further methods to subtract the likely distortions in the query at both audio level and after processing at signature and feature level are described.

One or more electronic devices include one or more processors that process different segments or channels of an audio signal of sound for a user. The one or more processors process a first segment or channel of the sound with first HRTFs so the first segment or channel externally localizes in front of and to a right of a head of the user and process a second segment or channel of the sound with second HRTFs so the second segment or channel externally localizes in front of and to a left of the head of the user.

One or more electronic devices include one or more processors that process different segments or channels of an audio signal of sound for a user. The one or more processors process a first segment or channel of the sound with first HRTFs so the first segment or channel externally localizes in front of and to a right of a head of the user and process a second segment or channel of the sound with second HRTFs so the second segment or channel externally localizes in front of and to a left of the head of the user.

An audio signal processing method and apparatus for adaptively adjusting a decorrelator. The method comprises obtaining a control parameter and calculating mean and variation of the control parameter. Ratio of the variation and mean of the control parameter is calculated, and a decorrelation parameter is calculated based on the said ratio. The decorrelation parameter is then provided to a decorrelator.

In accordance with embodiments of the present disclosure, a method for processing audio information in an audio device may include reproducing audio information by generating an audio output signal for communication to at least one transducer of the audio device, receiving at least one input signal indicative of ambient sound external to the audio device, detecting from the at least one input signal a near-field sound in the ambient sound, and modifying a characteristic of the audio information reproduced to the at least one transducer in response to detection of the near-field sound.

Embodiments of the present application provide an audio processing method and an apparatus. The method includes: a mobile terminal and a vehicle terminal are in a connected state, and playing, by the mobile terminal, a first audio synchronously with the vehicle terminal; obtaining, by the mobile terminal, a recorded audio of a current environment, where the recorded audio includes the first audio played by the vehicle terminal and a second audio for voice recognition; and eliminating, according to the first audio played by the mobile terminal, the first audio played by the vehicle terminal in the recorded audio to obtain the second audio. In the embodiments of the present application, by playing the first audio synchronously by the mobile terminal and the vehicle terminal, the second audio for voice recognition in the recorded audio can be obtained according to the first audio played by the mobile terminal.

Embodiments of the present application provide an audio processing method and an apparatus. The method includes: a mobile terminal and a vehicle terminal are in a connected state, and playing, by the mobile terminal, a first audio synchronously with the vehicle terminal; obtaining, by the mobile terminal, a recorded audio of a current environment, where the recorded audio includes the first audio played by the vehicle terminal and a second audio for voice recognition; and eliminating, according to the first audio played by the mobile terminal, the first audio played by the vehicle terminal in the recorded audio to obtain the second audio. In the embodiments of the present application, by playing the first audio synchronously by the mobile terminal and the vehicle terminal, the second audio for voice recognition in the recorded audio can be obtained according to the first audio played by the mobile terminal.

A pattern recognition apparatus includes: a model storage part that stores a model(s) generated by using transfer path information indicating a difference of transfer paths of a signal(s) for training, additional to the signal(s) for training, and a pattern recognition part that inputs an input signal and transfer path information indicating a difference of transfer paths of the input signal, and performs pattern recognition of the input signal by using the model(s).