Qualcomm Incorporated

Gwangju Institute of Science and Technology

Samsung Electronics Co., Ltd.

Electronics and Telecommunications Research Institute

Jongwon Shin

Erik Visser

Lae-Hoon Kim

Sang-Uk Ryu

Andre Gustavo P Schevciw

Yinyi Guo

Ian Ernan Liu

Tae Jin Lee

Seung Kwon Beack

Khaled Helmi El-Maleh

Samir Kumar Gupta

Pei Xiang

Te-Won Lee

Jongmo Sung

Kwokleung Chan

Woo-taek Lim

Ren Li

Hyun Jin Park

Inseon Jang

Jeremy Patrick Toman

Asif Mohammad

Seongwoo Kim

Roy Benjamin Silverstein

Heejong Yoo

Dongil Hyun

Youngju Cheon

Soojoong Hwang

Sangwook Han

Junhyeong Pak

Mofei Zhu

Sein Cheong

Asif Iqbal Mohammad

An audio encoding/decoding apparatus and method using vector quantized residual error features are disclosed. An audio signal encoding method includes outputting a bitstream of a main codec by encoding an original signal, decoding the bitstream of the main codec, determining a residual error feature vector from a feature vector of a decoded signal and a feature vector of the original signal, and outputting a bitstream of additional information by encoding the residual error feature vector.

Audio encoding/decoding apparatus and method using vector quantized residual error feature

A speech processing apparatus includes a plurality of microphones configured to receive a plurality of input signals, and processing circuitry configured to generate a spatial filtering signal corresponding to the plurality of input signals through spatial filtering, generate estimated noise information by integrating directional noise information representing a level of a noise signal received from a direction of interest with diffuse noise information representing levels of noise signals received from various directions based on whether the plurality of input signals have directionality, and generate an estimated speech signal by filtering the spatial filtering signal based on the estimated noise information.

Speech processing apparatus and method using a plurality of microphones

Disclosed herein is a method of enhancing speech. The method includes calculating a far-end speech spectrum by performing fast Fourier transformation of a signal received by a far-end user, calculating a background noise spectrum collected by a microphone provided to a mobile device of a near-end user; calculating a gain from the far-end speech spectrum and the background noise spectrum using a speech intelligibility index-based module, and deriving an enhanced far-end speech spectrum by applying the gain to the far-end speech spectrum, wherein, in calculating a gain using a speech intelligibility index-based module, a power budget used for transmitting and receiving a speech signal is set to vary with the background noise spectrum.

Method of enhancing speech using variable power budget

Disclosed is an application interface that takes into account the user's gaze direction relative to who is speaking in an interactive multi-participant environment where audio-based contextual information and/or visual-based semantic information is being presented. Among these various implementations, two different types of microphone array devices (MADs) may be used. The first type of MAD is a steerable microphone array (a.k.a. a steerable array) which is worn by a user in a known orientation with regard to the user's eyes, and wherein multiple users may each wear a steerable array. The second type of MAD is a fixed-location microphone array (a.k.a. a fixed array) which is placed in the same acoustic space as the users (one or more of which are using steerable arrays).

Audio user interaction recognition and application interface

A system which tracks a social interaction between a plurality of participants, includes a fixed beamformer that is adapted to output a first spatially filtered output and configured to receive a plurality of second spatially filtered outputs from a plurality of steerable beamformers. Each steerable beamformer outputs a respective one of the second spatially filtered outputs associated with a different one of the participants. The system also includes a processor capable of determining a similarity between the first spatially filtered output and each of the second spatially filtered outputs. The processor determines the social interaction between the participants based on the similarity between the first spatially filtered output and each of the second spatially filtered outputs.

Audio user interaction recognition and context refinement

A personalized (i.e., speaker-derivable) bandwidth extension is provided in which the model used for bandwidth extension is personalized (e.g., tailored) to each specific user. A training phase is performed to generate a bandwidth extension model that is personalized to a user. The model may be subsequently used in a bandwidth extension phase during a phone call involving the user. The bandwidth extension phase, using the personalized bandwidth extension model, will be activated when a higher band (e.g., wideband) is not available and the call is taking place on a lower band (e.g., narrowband).

Personalized bandwidth extension

A method for signal level matching by an electronic device is described. The method includes capturing a plurality of audio signals from a plurality of microphones. The method also includes determining a difference signal based on an inter-microphone subtraction. The difference signal includes multiple harmonics. The method also includes determining whether a harmonicity of the difference signal exceeds a harmonicity threshold. The method also includes preserving the harmonics to determine an envelope. The method further applies the envelope to a noise-suppressed signal.

Systems and methods for audio signal processing

Apparatus and methods for audio noise attenuation are disclosed. An audio signal analyzer can determine whether an input audio signal received from a microphone device includes a noise signal having identifiable content. If there is a noise signal having identifiable content, a content source is accessed to obtain a copy of the noise signal. An audio canceller can generate a processed audio signal, having an attenuated noise signal, based on comparing the copy of the noise signal to the input audio signal. Additionally or alternatively, data may be communicated on a communication channel to a separate media device to receive at least a portion of the copy of the noise signal from the separate media device, or to receive content-identification data corresponding to the content source.

Content based noise suppression

Implementations and applications are disclosed for detection of a transition in a voice activity state of an audio signal, based on a change in energy that is consistent in time across a range of frequencies of the signal. For example, such detection may be based on a time derivative of energy for each of a number of different frequency components of the signal.

Systems, methods, and apparatus for speech feature detection

A mobile device that is capable of automatically starting and ending the recording of an audio signal captured by at least one microphone is presented. The mobile device is capable of adjusting a number of parameters related with audio logging based on the context information of the audio input signal.

System and method of smart audio logging for mobile devices

A method of decomposing a multichannel audio signal is described. The method includes, for each of a plurality of frequency components of a segment in time of the multichannel audio signal, calculating a corresponding indication of a direction of arrival. The method also includes, based on the calculated direction indications, selecting a subset of the plurality of frequency components. The method further includes, based on the selected subset and on a plurality of basis functions, calculating a vector of activation coefficients. Each activation coefficient of the vector corresponds to a different basis function of the plurality of basis functions.

Systems, method, apparatus, and computer-readable media for decomposition of a multichannel music signal

Methods and apparatus for generating an anti-noise signal and equalizing a reproduced audio signal (e.g., a far-end telephone signal) are described, wherein the generating and the equalizing are both based on information from an acoustic error signal.

Systems, methods, devices, apparatus, and computer program products for audio equalization

Systems, methods, apparatus, and machine-readable media for voice activity detection in a single-channel or multichannel audio signal are disclosed.

Systems, methods, and apparatus for voice activity detection

An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

Growing community of inventors

Gwangju, South Korea

Jongwon Shin