Abstract: A wearable computer is configured to use beamforming techniques to isolate a user's speech from extraneous audio signals occurring within a physical environment. A microphone array of the wearable computer may generate audio signal data from an utterance from a user's mouth. A motion sensor(s) of the wearable computer may generate motion data from movement of the wearable computer. This motion data may be used to determine a direction vector pointing from the wearable computer to the user's mouth, and a beampattern may be defined that has a beampattern direction in substantial alignment with the determined direction vector to focus the microphone array on the user's mouth for speech isolation.

Abstract: Embodiments of the present disclosure can provide systems, methods, and computer-readable medium for providing audio and/or video effects based at least in part on facial features and/or voice feature characteristics of the user. For example, video and/or an audio signal of the user may be recorded by a device. Voice audio features and facial feature characteristics may be extracted from the voice audio signal and the video, respectively. The facial features of the user may be used to modify features of a virtual avatar to emulate the facial feature characteristics of the user. The extracted voice audio features may modified to generate an adjusted audio signal or an audio signal may be composed from the voice audio features. The adjusted/composed audio signal may simulate the voice of the virtual avatar. A preview of the modified video/audio may be provided at the user's device.

Abstract: An apparatus, system, and method allow a user to initiate a live stream that includes retrospectively captured images. Initiating the live stream includes providing provisional data that includes retrospectively captured images to a wireless radio.

Abstract: Provided is an apparatus and method for encoding/decoding audio based on a block. A method of encoding an audio signal may include dividing each of frame of input signal that constitute an audio signal into a plurality of subframes; transforming the subframes to a frequency domain; determining a two-dimensional (2D) intra block using the subframes transformed to the frequency domain; and encoding the 2D intra block. The 2D intra block may be a block that two-dimensionally displays frequency coefficients of the subframes transformed to the frequency domain using a time and a frequency.

Abstract: A UE includes an EUTRA-CMR reception unit that receives a codec mode request (EUTRA-CMR) including a codec mode that is determined by an eNB in accordance with a radio condition of the UE, a mode switching notification unit that notifies an encoder of switching to the codec mode included in the received codec mode request; and a mode switching acknowledgement unit that transmits a response message to the eNB when confirming that the encoder switches the codec mode.

Abstract: Systems and methods for text normalization in a plurality of noisy channels receive a text entry and channel origin data of the text entry; determine whether the text entry matches an in-vocabulary (IV) entry or whether the text entry is an out-of-vocabulary (OOV) entry; if the text entry is determined to have a matching IV entry, output the matching IV entry, and if the text entry is determined to be an OOV entry, implement a channel-specific error-type adapter framework based on the channel origin data, wherein the channel-specific error-type adapter framework is optimized for a specific channel from which the text entry originated; normalize the text entry using the channel-specific error-type adapter framework; and output one or more candidate normalized forms of the text entry.

Abstract: The present invention relates to a collision force analysis (CFA) method for obtaining in vivoestimates of contact force and pressure in the vocal cords directly from laryngeal videoendoscopies. The method comprises the steps of: providing at least a high-speed laryngeal videoendoscopy (also called High Speed Videoendoscopy or HSV) to generate at least one image and videos of the vocal cords; pre-processing the image in a processing unit, to define a region of interest (ROI) of the location of the glottis; detecting in the processing unit, the edges of the vocal folds in the images obtained by means of the HSV; recording in the processing unit, the points of the edges detected by means of a sequence of images; estimating in the processing unit, the path of the vocal cord edge during collision throughout time; and estimating the values of contact and impact of the vocal cords by means of a collision model.

Abstract: There is described a method for evaluating a level of noise in a biosignal, the method comprising: receiving a time signal representative of a biological activity, the time signal comprising a biological activity component and a noise component; determining a modulation spectrum for the time signal, the modulation spectrum representing a signal frequency as a function of a modulation frequency; from the modulation spectrum determining a first amount of modulation energy corresponding to the biological activity component and a second amount of modulation energy corresponding to the noise component determining an indication of the level of noise using the first and second amounts of modulation energy; and outputting the indication of the level of noise.

Abstract: An information processing apparatus identifies, by using an audio signal acquired by collecting a user's voice, evaluation target time that includes at least either time not including the user's voice or time during which the user is producing a meaningless utterance and produces an output appropriate to the identified evaluation target time.

Abstract: Method, apparatus and systems for object tracking are disclosed. In one example, a system for tracking a plurality of objects in a venue is disclosed. The system comprises a transmitter configured for transmitting a series of probe signals in a broadcasting manner through a wireless multipath channel, wherein the wireless multipath channel is impacted by a movement of at least one of the plurality of objects in the venue; and a plurality of heterogeneous target wireless receivers each of which is associated with an object of the plurality of objects in the venue.

Abstract: A first signal weighting processor outputs a weighted signal obtained by performing a weighting on part of an input signal representing a feature of a target signal included in the input signal. A neural network processor outputs an enhancement signal for the target signal by using a coupling coefficient. An inverse filter cancels the weighting on the feature representation of the target signal in the enhancement signal. A second signal weighting processor outputs a weighted signal obtained by performing a weighting on part of a supervisory signal representing a feature of a target signal. An error evaluator output a coupling coefficient to have a value indicating that a learning error between the weighted signal output from the second signal weighting processor and the output signal of the neural network processor is less than or equal to a set value.

Abstract: A computer-implemented method for management of voicemail messages, performed at a portable electronic device with a touch screen display, includes: displaying a list of voicemail messages; detecting selection by a user of a respective voicemail message in the list; responding to the user selection of the respective voicemail message by initiating playback of the user-selected voicemail message; displaying a progress bar for the user-selected voicemail message, wherein the progress bar indicates the portion of the user-selected voicemail message that has been played; detecting movement of a finger of the user from a first position on the progress bar to a second position on the progress bar; and responding to the detection of the finger movement by restarting playback of the user-selected voicemail message at a position within the user-selected voicemail message corresponding substantially to the second position on the progress bar.

Abstract: A codec allowing for switching between different coding modes is improved by, responsive to a switching instance, performing temporal smoothing and/or blending at a respective transition.

Abstract: A method, computer system, and a computer program product for generating a plurality of voice data having a particular speaking style is provided. The present invention may include preparing a plurality of original voice data corresponding to at least one word or at least one phrase is prepared. The present invention may also include attenuating a low frequency component and a high frequency component in the prepared plurality of original voice data. The present invention may then include reducing power at a beginning and an end of the prepared plurality of original voice data. The present invention may further include storing a plurality of resultant voice data obtained after the attenuating and the reducing.

Abstract: An online system authenticates a user through a voiceprint biometric verification process. When a user needs to be authenticated, the online system generates and provides a random phrase to the user. The online system receives an audio recording of the randomly generated phrase and retrieves a previously trained voiceprint model for the user. The online system analyzes the audio recording by applying the voiceprint model to determine whether the audio recording satisfies a first criteria of whether the voice in the audio recording belongs the user and a second criteria of whether the audio recording includes a vocalization of the randomly generated phrase. If the audio recording satisfies both criteria, the online system authenticates the user. Therefore, the user can be provided access to a new communication session in response to being authenticated.

Abstract: Tokens are assigned to data to be processed. Each token is associated with a unique sound waveform defined by at least two contemporaneous tones of different frequency. The waveforms can be emitted as sound at computing devices having speakers. Other computing devices have microphones and capture emitted sound to extract tokens from the waveforms. Extracted tokens can be taken as confirmation that the data is to be processed. Data can represent transactions.

Abstract: An apparatus includes a capture device and a processor. The capture device may be configured to generate video frames of an environment near a vehicle. The processor may be configured to receive video frames from the capture device, perform video operations to detect objects in the video frames, determine advertisement locations in the video frames based on the objects detected in the video frames, insert advertisements at the advertisement locations and generate video data for a display. The video data may comprise the video frames with the inserted advertisements.

Abstract: An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided, having: a receiving interface for receiving one or more frames, a coefficient generator, and a signal reconstructor. The coefficient generator is configured to determine one or more first audio signal coefficients, and one or more noise coefficients. Moreover, the coefficient generator is configured to generate one or more second audio signal coefficients, depending on the one or more first audio signal coefficients and depending on the one or more noise coefficients.

Abstract: A system, method and program product for generating sound masking in an open-plan space. A method is disclosed that includes: establishing a set of acoustic criteria for the space that specifies minimum output levels at a set of specified frequencies; mixing sound samples to create an audio output stream for use as sound masking when broadcast over at least one speaker using a power amplifier; and processing the audio output stream, wherein the processing includes: analyzing the audio output stream with a spectrum analyzer to determine if the minimum output levels at the set of specified frequencies are met; and level adjusting the audio output stream with an equalizer to ensure that minimum output levels at the set of specified frequencies are met but do not exceed a given threshold above the minimum level.

Abstract: Techniques are generally described for enhanced compression of video data. In various examples, the techniques may include receiving first video data representing a scene in an environment. The techniques may further include generating illumination map data representing illumination of the scene in the first video data. The techniques may further comprise generating reflectance map data representing a reflectance of at least one object in the first video data. In some examples, the techniques may include sending, to a second computing device, the illumination map data and the reflectance map data. The techniques may further include receiving second video data representing the scene. The techniques may include determining a first illumination difference between the second video data and the first video data. The techniques may comprise sending, to the second computing device, the first illumination difference.

Abstract: Systems and methods are disclosed configured to train an autoencoder using images that include faces, wherein the autoencoder comprises an input layer, an encoder configured to output a latent image from a corresponding input image, and a decoder configured to attempt to reconstruct the input image from the latent image. An image sequence of a face exhibiting a plurality of facial expressions and transitions between facial expressions is generated and accessed. Images of the plurality of facial expressions and transitions between facial expressions are captured from a plurality of different angles and using different lighting. An autoencoder is trained using source images that include the face with different facial expressions captured at different angles with different lighting, and using destination images that include a destination face.

Abstract: Aspects of the disclosure include an apparatus that has a first clock generator and a second clock generator. The first clock generator is configured to drive a first circuit, causing the first circuit to (i) receive a signal corresponding to an audio input, and (ii) determine whether an energy level of the signal exceeds a predetermined threshold. The second clock generator is activated when the first circuit determines that the energy level of the signal exceeds the predetermined threshold. The second clock generator is configured to drive a second circuit, causing the second circuit to determine whether the signal matches a predetermined pattern. A third circuit is activated when the second circuit determines that the signal matches the predetermined pattern.

Abstract: A cache associated with a voice recognition engine may be configured to receive an audio file corresponding to a voice query. The voice query may be received by the voice recognition engine from a user device in response to receipt of the voice query by the user device at the user device. The cache may be configured to determine an audio fingerprint based on the audio file and to determine whether the audio fingerprint corresponds to one of a plurality of audio transcriptions stored in the cache. The audio transcriptions may correspond to popular voice queries received at the voice recognition engine and processed by a server capable of performing automated speech recognition. In response to determining that the audio fingerprint corresponds to a given one of the stored audio transcriptions, the audio file may be processed based on the stored audio transcription.

Abstract: A method of determining a distribution of bits for coding a transition frame, said method being implemented in a coder/decoder for coding/decoding a digital signal, the transition frame being preceded by a predictive coded preceding frame, coding the transition frame comprising transform coding and predictive coding a single sub-frame of the transition frame, the method comprising the following steps: assigning a bit rate for predictive coding the transition sub-frame, said bit rate being equal to the minimum between the bit rate for transform coding the transition frame and a first predetermined bit rate value; determining a first number of bits allocated for predictive coding the transition sub-frame for said bit rate; and calculating a second number of bits allocated for transform coding the transition frame from the first number of bits and a number of bits available for coding the transition frame.

Abstract: A system for testing line attenuation defects includes a data transmission line configured to transmit a forward signal in a first direction, at least one reflection point at a first location along the data transmission line, a test probe configured to (i) electrically contact a center conductor at a second location along the data transmission line, (ii) introduce a broadband data signal onto the data transmission line, and (iii) measure, at the second location, a return signal from the reflection point, and a spectrum capturing device in operable contact with the test probe. The spectrum capturing device is configured to (i) collect and arrange frequency data measured by the test probe for the test signal, the return signal, and a standing wave created by the sum of the broadband data signal and the return signal, (ii) determine the voltage VSWR of the standing wave, and (iii) calculate a line loss from the VSWR.

Abstract: An apparatus and a method to encode and decode a speech signal using an encoding mode are provided. An encoding apparatus may select an encoding mode of a frame included in an input speech signal, and encode a frame having an unvoiced mode for an unvoiced speech as the selected encoding mode.

Abstract: A method for acquiring the number of modified frames for active sound, and a method and apparatus for voice activity detection are disclosed. Firstly, a first voice activity detection decision result and a second voice activity detection decision result are obtained (501), the number of hangover frames for active sound is obtained (502), and the number of background noise updates is obtained (503), and then the number of modified frames for active sound is calculated according to the first voice activity detection decision result, the number of background noise updates and the number of hangover frames for active sound (504), and finally, a voice activity detection decision result of a current frame is calculated according to the number of modified frames for active sound and the second voice activity detection decision result (505).

Abstract: A method of encoding samples in a digital signal is provided that includes receiving a frame of N samples of the digital signal, determining L possible distinct data values in the N samples, determining a reference data value in the L possible distinct data values and a coding order of L?1 remaining possible distinct data values, wherein each of the L?1 remaining possible distinct data values is mapped to a position in the coding order, decomposing the N samples into L?1 coding vectors based on the coding order, wherein each coding vector identifies the locations of one of the L?1 remaining possible distinct data values in the N samples, and encoding the L?1 coding vectors.

Abstract: The present invention relates to an audio encoding and, more particularly, to a signal classifying method and device, and an audio encoding method and device using the same, which can reduce a delay caused by an encoding mode switching while improving the quality of reconstructed sound. The signal classifying method may comprise the operations of: classifying a current frame into one of a speech signal and a music signal; determining, on the basis of a characteristic parameter obtained from multiple frames, whether a result of the classifying of the current frame includes an error; and correcting the result of the classifying of the current frame in accordance with a result of the determination. By correcting an initial classification result of an audio signal on the basis of a correction parameter, the present invention can determine an optimum coding mode for the characteristic of an audio signal and can prevent frequent coding mode switching between frames.

Abstract: The application discloses a lip movement capturing method and device and a storage medium. The method includes: acquiring a real-time image shot by a photographic device and extracting a real-time facial image from the real-time image; inputting the real-time facial image into a pretrained lip average model and recognizing t lip feature points representative of positions of lips in the real-time facial image; and calculating a movement direction and movement distance of the lips in the real-time facial image according to x and y coordinates of the t lip feature points in the real-time facial image. According to the application, movement information of the lips in the real-time facial image is calculated according to the coordinates of the lip feature points to implement real-time capturing of movements of the lips.

Abstract: An apparatus for encoding one or more audio objects to obtain an encoded signal is provided. The apparatus includes a for downmixing the one or more audio objects to obtain one or more unprocessed downmix signals. Moreover, the apparatus includes a processing module and a signal calculator. The signal calculator is configured to calculate each of one or more additional signals based on a difference between one of one or more processed downmix signals and one of the one or more unprocessed downmix signals. Moreover, the apparatus includes an object information generator. Furthermore, the apparatus includes an output interface for outputting the encoded signal. Moreover, a corresponding apparatus for decoding is provided.

Abstract: A receiver for demodulating a pulse width modulated (“PWM”) signal, comprises: a voltage level shifter for shifting the PWM signal to predefined transistor voltage levels; a half-rate PWM decoder for receiving the shifted PWM signal; and a 2-bit-to-N-bit deserializer. The half-rate PWM decoder comprises a first decoder core, a second decoder core, a controller, and a sampler and retiming circuit. The first decoder core and the second decoder core are configured to decode alternating periods of the shifted PWM signal. The controller is coupled to the first decoder core, the second decoder core, the sampler and retiming circuit. The retiming circuit is configured to receive clock signals from the controller and to output half-rate even data from the first decoder core and half-rate odd data from the second decoder core. Outputs of the retiming circuit and an output of the controller are coupled to inputs of the deserializer.

Abstract: Provided are a method and an apparatus for determining an encoding mode for improving the quality of a reconstructed audio signal. A method of determining an encoding mode includes determining one from among a plurality of encoding modes including a first encoding mode and a second encoding mode as an initial encoding mode in correspondence to characteristics of an audio signal, and if there is an error in the determination of the initial encoding mode, generating a modified encoding mode by modifying the initial encoding mode to a third encoding mode.

Abstract: According to one embodiment, a signal processing apparatus includes a memory and a processor electrically coupled to the memory. The processor separates a plurality of signals by a separation filter, and outputs a plurality of separate signals. The plurality of signals includes signals which are received at different positions and come from different directions. The processor estimates incoming directions of the plurality of separate signals, respectively, and associates the plurality of separate signals with transmission sources of the signals, and present association between the plurality of separate signals and the transmission sources of the signals.

Abstract: An electronic device including an earphone device is provided. The earphone device includes a shell, a speaker, a first microphone device, a memory circuit and a controller. The memory circuit stores multiple parameter sets. The first microphone device receives a first sound. The first microphone device generates first data based on the first sound. The controller compares the first data with the parameter sets of the memory circuit and determines which one of the parameter sets corresponds to the first data based on the frequency parameters and the volume parameters. The controller generates second data based on the adjustment parameters of the one of the parameter sets, and the speaker generates a second sound based on the second data. The first sound generates a third sound in the shell, and the phase of the second sound is substantially opposite to the phase of the third sound.

Abstract: A device for sound wave communication, including: a hardware correction table for setting a correction frequency band; a sound wave transmission unit for generating data frequencies at a predetermined base decibel level, generating separate reception filter frequencies at the base decibel level for receiving data carried by a sound wave transmitted from the nearest location when sound waves are received, and generating the correction reference frequency at the base decibel level for correcting hardware transmission; and a sound wave reception unit for receiving a sound wave signal transmitted from the sound wave transmission unit, extracting decibel levels at each of the data frequencies to form an array of decibel levels, correcting the array by shifting the array by a correction value extracted using the hardware correction table, and reconstructing data.

Abstract: An embodiment of the invention provides a wireless ear-borne audio device that may be configured in a variety of ways, including, but in no way limited to a device for recording audio information and storing the audio information for later replay, and/or forwarding the audio information to another device. The audio information may be analyzed to perform further functions. An embodiment of the invention also provides a method for compressing audio data and transmitting the audio information to the ear-borne audio device in a manner that reduces the power consumption of the ear-borne audio device in receiving data via a Bluetooth® connection.

Abstract: Provided is a communication method for supporting various bandwidths. The device detects a synchronization signal transmitted by a wide-band cell and receives a downlink (DL) signal from the wide-band cell on the basis of the synchronization signal. A support bandwidth supported by the device is narrower than a system bandwidth supported by the wide-band cell.

Abstract: Aspects of the present disclosure provide techniques for compressing data packets for cellular internet of things (CIoT) communications. An example method generally includes establishing at least one prefill buffer common to one or more UEs, wherein the prefill buffer includes a plurality of common strings, generating a compressed packet by finding matches to the common strings in at least one of a header portion or payload portion of the packet and associating identifiers with the common strings, and transmitting the packet.

Abstract: A first example playback device includes one or more processors and a non-transitory computer-readable medium storing instructions that, when executed by the one or more processors, cause the first playback device to perform functions. The functions include generating a mixed audio signal comprising one or more components of a first audio stream and one or more components of a second audio stream; and playing the mixed audio signal, where the one or more components of the second audio stream of the mixed audio signal played by the first playback device are played in synchrony with a second playback device configured to play the second audio stream. An example non-transitory computer-readable medium and an example method, both related to the first example playback device, are also disclosed herein.

Abstract: The invention provides methods and devices for outputting a stereo audio signal having a left channel and a right channel. The apparatus includes a demultiplexer, decoder, and upmixer. The upmixer is configured operate either in a prediction mode or a non-prediction mode based on a parameter encoded in the audio bitstream.

Abstract: A method and device are provided for determining an optimized scale factor to be applied to an excitation signal or a filter during a process for frequency band extension of an audio frequency signal. The band extension process includes decoding or extracting, in a first frequency band, an excitation signal and parameters of the first frequency band including coefficients of a linear prediction filter, generating an excitation signal extending over at least one second frequency band, filtering using a linear prediction filter for the second frequency band. The determination method includes determining an additional linear prediction filter, of a lower order than that of the linear prediction filter of the first frequency band, the coefficients of the additional filter being obtained from the parameters decoded or extracted from the first frequency and calculating the optimized scale factor as a function of at least the coefficients of the additional filter.

Abstract: The disclosure provides a noise reduction method and an electronic device. In an embodiment of the disclosure, when determining that a plurality of first applications occupy a plurality of first audio channels connected with a microphone and a second application occupies a second audio channel connected with a speaker, the electronic device resamples the audio data of the second audio channel according to the sampling rates corresponding to the plurality of first audio channels and then performs the noise reduction processing on the audio data of each of the plurality of first audio channels respectively according to the audio data obtained by resampling.

Abstract: An audio encoding method and an apparatus are provided. The method includes: determining sparseness of distribution, on spectrums, of energy of N input audio frames (101), where the N audio frames include a current audio frame, and N is a positive integer; and determining, according to the sparseness of distribution, on the spectrums, of the energy of the N audio frames, whether to use a first encoding method or a second encoding method to encode the current audio frame (102), where the first encoding method is an encoding method that is based on time-frequency transform and transform coefficient quantization and that is not based on linear prediction, and the second encoding method is a linear-predication-based encoding method. The method can reduce encoding complexity and ensure that encoding is of relatively high accuracy.

Abstract: Concealing a lost audio frame of a received audio signal by performing a sinusoidal analysis of a part of a previously received or reconstructed audio signal, wherein the sinusoidal analysis involves identifying frequencies of sinusoidal components of the audio signal, applying a sinusoidal model on a segment of the previously received or reconstructed audio signal, wherein said segment is used as a prototype frame in order to create a substitution frame for a lost audio frame, and creating the substitution frame for the lost audio frame by time-evolving sinusoidal components of the prototype frame, up to the time instance of the lost audio frame, in response to the corresponding identified frequencies.

Abstract: In described examples, a method for detecting voice activity includes: receiving a first input signal containing noise; sampling the first input signal to form noise samples; determining a first value corresponding to the noise samples; subsequently receiving a second input signal; sampling the second input signal to form second signal samples; determining a second value corresponding to the second signal samples; forming a ratio of the second value to the first value; comparing the ratio to a predetermined threshold value; and responsive to the comparing, indicating whether voice activity is detected in the second input signal.

Abstract: Enveloping techniques using incoherent wavefront multiplexing (WF muxing or K-muxing) will enhance privacy protection on data communications. The disclosure relates to methods and architectures of packing or enveloping data using WF muxing, or K-muxing, for information transport via multiple communication links such as concurrently via multiple satellites, airborne platforms, wireless terrestrial links, and/or other wireless links. The multi-link communications may include the use of cloud transport of multiple WF-muxed data packages. It is focused to appearance of a digital envelop and reliability of enclosed data. The K-muxing on information digital streams before modulation in a transmitter shall provide enhanced data privacy and better availability. The WF multiplexed (WF muxed or K-muxed) information data streams will be individually and concurrently sent to the multiple links accordingly for data transport.

Abstract: A signal acquisition probe stores compressed or compressed and filtered time domain data samples representing at least one of an impulse response or step response characterizing the signal acquisition probe. The compressed or compressed and filtered time domain data samples of the impulse response or the step response are provided to a signal measurement instrument for compensating the signal measurement instrument for the impulse or step response of the signal measurement instrument.

Abstract: Provided are a frame error concealment method and apparatus and an error concealment scheme construction method and apparatus. The frame error concealment method includes generating a new signal by synthesizing a plurality of previous signals that are similar to a signal of an error frame and reconstructing the signal of the error frame using the generated signal.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part.