Abstract: An apparatus for generating a bandwidth enhanced audio signal from an input audio signal having an input audio signal frequency range includes: a raw signal generator configured for generating a raw signal having an enhancement frequency range, wherein the enhancement frequency range is not included in the input audio signal frequency range; a neural network processor configured for generating a parametric representation for the enhancement frequency range using the input audio frequency range of the input audio signal and a trained neural network; and a raw signal processor for processing the raw signal using the parametric representation for the enhancement frequency range to obtain a processed raw signal having frequency components in the enhancement frequency range, wherein the processed raw signal or the processed raw signal and the input audio signal frequency range of the input audio signal represent the bandwidth enhanced audio signal.

Abstract: Speech recognition may be improved by generating and using a topic specific language model. A topic specific language model may be created by performing an initial pass on an audio signal using a generic or basis language model. A speech recognition device may then determine topics relating to the audio signal based on the words identified in the initial pass and retrieve a corpus of text relating to those topics. Using the retrieved corpus of text, the speech recognition device may create a topic specific language model. In one example, the speech recognition device may adapt or otherwise modify the generic language model based on the retrieved corpus of text.

Abstract: An audio signal encoding method performed by an encoder includes identifying a time-domain audio signal in a unit of blocks, quantizing a linear prediction coefficient extracted from a combined block in which a current original block of the audio signal and a previous original block chronologically adjacent to the current original block using frequency-domain linear predictive coding (LPC), generating a temporal envelope by dequantizing the quantized linear prediction coefficient, extracting a residual signal from the combined block based on the temporal envelope, quantizing the residual signal by one of time-domain quantization and frequency-domain quantization, and transforming the quantized residual signal and the quantized linear prediction coefficient into a bitstream.

Abstract: An encoder operable to filter audio signals into a plurality of frequency band components, generate quantized digital components for each band, identify a potential for pre-echo events within the generated quantized digital components, generate an approximate signal by decoding the quantized digital components using inverse pulse code modulation, generate an error signal by comparing the approximate signal with the sampled audio signal, and process the error signal and quantized digital components. The encoder operable to process the error signal by processing delayed audio signals and Q band values, determining the potential for pre-echo events from the Q band values, and determining scale factors and MDCT block sizes for the potential for pre-echo events.

Abstract: Provided is a system which allows a learner who is a non-native speaker of a given language to intuitively improve pronunciation of the language. A pronunciation conversion apparatus includes a conversion section which converts a first feature value corresponding to a first speech signal obtained when a first speaker who speaks a given language as his/her native language speaks another language such that the first feature value approaches a second feature value corresponding to a second speech signal obtained when a second speaker who speaks the other language as his/her native language speaks the other language, each of the first feature value and the second feature value is a feature value capable of representing a difference in pronunciation, and a speech signal obtained from the first feature value after the conversion is presented to the first speaker.

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.