Abstract: This encoding device is able to encode an S signal efficiently in MS prediction encoding. An M signal encoding unit generates first encoding information by encoding a sum signal indicating a sum of a left channel signal and a right channel signal that constitute a stereo signal. An energy difference calculation unit calculates a prediction parameter for predicting a difference signal indicating a difference between the left channel signal and the right channel signal by using a parameter regarding an energy difference between the left channel signal and the right channel signal. An entropy encoding unit generates second encoding information by encoding the prediction parameter.

Abstract: Provided is to prevent a false determination due to an attachment condition of an apparatus that transmits and receives an acoustic signal, and perform accurate personal authentication. A personal authentication device includes: a personal authentication means that authenticates an individual by using first information at least including an acoustic characteristic calculated from an acoustic signal propagating through the head of the user, which is detected by an apparatus being attached on a head of a user for transmitting and receiving the acoustic signal, and a feature amount extracted from the acoustic characteristic; an attachment trouble rule storage means that stores an attachment trouble rule for detecting an attachment trouble with the apparatus; and an attachment trouble detection means that detects a trouble with an attachment state of the apparatus when the first information satisfies the attachment trouble rule.

Abstract: Disclosed speech recognition techniques improve user-perceived latency while maintaining accuracy by: receiving an audio stream, in parallel, by a primary (e.g., accurate) speech recognition engine (SRE) and a secondary (e.g., fast) SRE; generating, with the primary SRE, a primary result; generating, with the secondary SRE, a secondary result; appending the secondary result to a word list; and merging the primary result into the secondary result in the word list. Combining output from the primary and secondary SREs into a single decoder as described herein improves user-perceived latency while maintaining or improving accuracy, among other advantages.

Abstract: Disclosed is a digital audio tampering forensics method based on phase offset detection, comprising: multiplying a signal to be identified with a time label to obtain a modulation signal of the signal to be identified; then, performing a short-time Fourier transform on the signal to be identified and the modulation signal to obtain a signal power spectrum and a modulation signal power spectrum; computing group delay characteristics by using the signal power spectrum and the modulation signal power spectrum; computing a mean value of the group delay characteristics, and then using the mean value results for smoothing computation to obtain phase information of a current frame signal; computing a dynamic threshold by using the phase information of the current frame signal, and then deciding whether the signal is tampered by using the dynamic threshold and the phase information of the current frame signal.

Abstract: The activities of multiple virtual personal assistant (VPA) applications are coordinated. For example, different portions of a conversational natural language dialog involving a user and a computing device may be handled by different VPAs.

Abstract: Provided are an electronic device and method for providing a voice assistant service. The method, performed by the electronic device, of providing the voice assistant service includes: obtaining a voice of a user; obtaining voice analysis information of the voice of the user by inputting the voice of the user to a natural language understanding model; determining whether a response operation with respect to the voice of the user is performable, according to a preset criterion, based on the obtained voice analysis information; and based on the determining that the response operation is not performable, outputting a series of guide messages for learning the response operation related to the voice of the user.

Abstract: One or more context aware processing parameters and an ambient audio stream are received. One or more sound characteristics associated with the ambient audio stream are identified using a machine learning model. One or more actions to perform are determined using the machine learning model and based on the one or more context aware processing parameters and the identified one or more sound characteristics. The one or more actions are performed.

Abstract: A system determines an event location of an event within an indoor environment based on an event sound generated by the event. The system employs time-reversal techniques based on a received event sound to identify the event location as being in the vicinity of one of a plurality of locator devices at locator locations in the environment. The system includes a base array located within the environment that receives an indication that an event has been detected. Upon receiving the event sound, the system generates a time-reversed event sound for each transceiver and transmits via each transceiver the time-reversed event sound for that transceiver. When a locator device receives a time-reversed event sound, the locator device determines whether the event is in the vicinity of that locator location of the locator device and, if so, outputs an indication that the event occurred at that locator location.

Abstract: A method of converting a frame of a voice sample to a singing frame includes obtaining a pitch value of the frame; obtaining formant information of the frame using the pitch value; obtaining aperiodicity information of the frame using the pitch value; obtaining a tonic pitch and chord pitches; using the formant information, the aperiodicity information, the tonic pitch, and the chord pitches to obtain the singing frame; and outputting or saving the singing frame.

Abstract: An event data recorder as a recording device for vehicles capable of recording clear sound data that may contribute to investigation of a cause of an accident in event recording data, regardless of a sound volume of an audio sound played back in the own vehicle, includes a video data acquisition unit configured to acquire video data including sound data and imaging an area around the own vehicle, a sound volume detection unit configured to detect a sound volume of an audio sound played back in the vehicle, and a recording control unit configured to record video data including data by which a sound volume of an audio sound can be determined and sound data on which an effect of an audio sound is small, when a sound volume of an audio sound detected by the sound volume detection unit is greater than a sound volume.

Abstract: Disclosed embodiments provide techniques for improved call quality during telephony sessions. The speech quality of an active voice session is periodically evaluated using multiple noise reduction algorithms. In an instance where the speech quality of the currently used noise reduction algorithm is below the quality of another noise reduction algorithm, the telephony system may switch to a new noise reduction algorithm as the currently used (active) noise reduction algorithm in order to improve call quality during an active voice session.

Abstract: Disclosed embodiments provide techniques for improved call quality during telephony sessions. The speech quality of an active voice session is periodically evaluated using multiple noise reduction algorithms. In an instance where the speech quality of the currently used noise reduction algorithm is below the quality of another noise reduction algorithm, the telephony system may switch to a new noise reduction algorithm as the currently used (active) noise reduction algorithm in order to improve call quality during an active voice session.

Abstract: A method is provided for acoustic source direction of arrival estimation and acoustic source separation, via spatial weighting of the dictionary based display of the steered response function calculated for a certain number of directions from spherical harmonic decomposition coefficients obtained from microphone array recordings of the sound field. The usage of spatial band limited functions of plane waves to represent more complex directional maps of the sound field constitutes the algorithm. These functions are calculated for pre-defined directions on an analysis surface (such as a sphere). The directions of arrival of sound sources are calculated with the same method in order to group source estimates to localize sound sources. Thereby, directions of arrival can be obtained from the recordings of the sound sources captured by means of a microphone array and following this, sound sources can be separated by using this direction information or predetermined source arrival directions.

Abstract: This application discloses a method and apparatus for processing audio information, a storage medium, and an electronic apparatus. The method includes: detecting that a segment of audio information is being received on a client, a first portion of audio information in the segment of audio information having been currently received on the client; obtaining first information, second information, and third information based on the first portion of audio information that has been currently received, the first information including text information corresponding to the first portion of audio information, the second information including information that meets a target condition and that corresponds to the first information, and the third information including information to be pushed to the client, which is obtained based on a keyword in the first information; and displaying the first information, the second information, and the third information on the client.

Abstract: There is disclosed inter alia an apparatus for spatial audio signal encoding which determines at least one spatial audio parameter comprising a direction parameter with an elevation component and an azimuth component. The elevation component and azimuth component of the direction parameter are then converted to an index value.

Abstract: In communication performed among multiple participants, at least one of a participant who will start speaking next and a timing thereof is estimated. An estimation apparatus includes a head motion information generation unit that acquires head motion information representing head motions of communication participants in a time segment corresponding to an end time of an utterance segment and synchronization information for head motions between the communication participants, and an estimation unit that estimates at least one of the speaker of the next utterance segment following the utterance segment and the next utterance start timing following the utterance segment based on the head motion information and the synchronization information for the head motions between the communication participants.

Abstract: This disclosure relates generally to a system and method to identify a plurality of noises or their combination to suppress them and enhancing the deteriorated input signal in a dynamic manner. It identifies noises in the audio signal and categorizing them based on the trained database of noises. A combination of deep neural network (DNN) and artificial Intelligence (AI) helps the system for self-learning to understand and capture noises in the environment and retain the model to reduce noises from the next attempt. The system suppresses unwanted noise coming from the external environment with the help of AI based algorithms, by understanding, differentiating, and enhancing human voice in a live environment. The system will help in the reduction of unwanted noises and enhance the experience of business and public meetings, video conferences, musical events, speech broadcasts etc. that could cause distractions, disturbances and create barriers in the conversation.

Abstract: A stereo signal encoding method includes obtaining a residual signal encoding parameter of a current frame of a stereo signal based on downmixed signal energy and residual signal energy of each of M sub-bands of the current frame, where the residual signal encoding parameter indicates whether to encode residual signals of the M sub-bands, determining whether to encode the residual signals based on the residual signal encoding parameter, and encoding the residual signals when it is determined that the residual signals need to be encoded.

Abstract: Methods, systems, and apparatus, including computer programs encoded on computer storage media, for audio-visual speech separation. A method includes: obtaining, for each frame in a stream of frames from a video in which faces of one or more speakers have been detected, a respective per-frame face embedding of the face of each speaker; processing, for each speaker, the per-frame face embeddings of the face of the speaker to generate visual features for the face of the speaker; obtaining a spectrogram of an audio soundtrack for the video; processing the spectrogram to generate an audio embedding for the audio soundtrack; combining the visual features for the one or more speakers and the audio embedding for the audio soundtrack to generate an audio-visual embedding for the video; determining a respective spectrogram mask for each of the one or more speakers; and determining a respective isolated speech spectrogram for each speaker.

Abstract: A system and method are described for automatic acoustic feedback cancellation in real time. In some implementations, the system may receive audio data describing an audio signal, which the system may use to determine a set of frames of the audio signal. Spectral analysis may be performed on the one or more frames of the audio to detect spectral patterns of two or more frames indicative of acoustic feedback. An additional delay identification test may be performed to identify a consistent delay indicative of acoustic feedback. In some implementations, a state machine is advanced based in part on accumulated delay votes. Decisions can be made to mute the acoustic feedback and cease the muting operation when silence is detected.