Abstract: Disclosed herein are example techniques to provide contextual information corresponding to a voice command. An example implementation may involve receiving voice data indicating a voice command, receiving contextual information indicating a characteristic of the voice command, and determining a device operation corresponding to the voice command. Determining the device operation corresponding to the voice command may include identifying, among multiple zones of a media playback system, a zone that corresponds to the characteristic of the voice command, and determining that the voice command corresponds to one or more particular devices that are associated with the identified zone. The example implementation may further involve causing the one or more particular devices to perform the device operation.

Abstract: A controller for a surgical navigation system is presented. The controller is configured to receive a position signal from a tracking system, wherein the position signal is indicative of a position of a hand-held surgical device that is tracked by the surgical navigation system inside an operation environment. The controller is further configured to receive sound signals from a plurality of microphones directed toward the operation environment, wherein the sound signals potentially contain one or more voice commands from one or more voice sources inside the operation environment. The controller is configured to process the sound signals dependent on the position signal.

Abstract: Audio decoder device for decoding a bitstream, the audio decoder device including: a predictive decoder for producing a decoded audio frame from the bitstream, wherein the predictive decoder includes a parameter decoder for producing one or more audio parameters for the decoded audio frame from the bitstream and wherein the predictive decoder includes a synthesis filter device for producing the decoded audio frame by synthesizing the one or more audio parameters for the decoded audio frame; a memory device including one or more memories, wherein each of the memories is configured to store a memory state for the decoded audio frame, wherein the memory state for the decoded audio frame of the one or more memories is used by the synthesis filter device for synthesizing the one or more audio parameters for the decoded audio frame; and a memory state resampling device configured to determine the memory state for synthesizing the one or more audio parameters for the decoded audio frame, which has a sampling rate, f

Abstract: A method and a device for reducing noise of a wireless-earphone, a wireless-earphone and a computer-readable storage medium are provided. In the method, a first electromagnetic signal from a slave earphone is received, and the first electromagnetic signal is converted to an audio signal. The first electromagnetic signal is obtained by a coil inducing an audio signal acquired by a microphone in the slave earphone. A to-be-denoised earphone is determined based on audio signals acquired by microphones in the earphones. It is determined whether an audio signal acquired by a microphone in the to-be-denoised earphone has a voice feature. Noise reduction processing is performed on the audio signal acquired by the microphone in the to-be-denoised earphone by using a filter in a case that the audio signal acquired by the microphone in the to-be-denoise earphone has the voice feature.

Abstract: Audio decoder device for decoding a bitstream, the audio decoder device including: a predictive decoder for producing a decoded audio frame from the bitstream, wherein the predictive decoder includes a parameter decoder for producing one or more audio parameters for the decoded audio frame from the bitstream and wherein the predictive decoder includes a synthesis filter device for producing the decoded audio frame by synthesizing the one or more audio parameters for the decoded audio frame; a memory device including one or more memories, wherein each of the memories is configured to store a memory state for the decoded audio frame, wherein the memory state for the decoded audio frame of the one or more memories is used by the synthesis filter device for synthesizing the one or more audio parameters for the decoded audio frame; and a memory state resampling device configured to determine the memory state for synthesizing the one or more audio parameters for the decoded audio frame, which has a sampling rate, f

Abstract: Provided are a method for estimating a spectrum density of diffused noises. Also provided is a processor for implementing the method. The processor includes at least two sound receiving units and a spectrum density estimating unit for estimating spectrum density.

Abstract: Embodiments of the present invention include methods and apparatuses for adjusting audio content when more multiple audio objects are directed toward a single audio output device. The amplitude, white noise content, and frequencies can be adjusted to enhance overall sound quality or make content of certain audio objects more intelligible. Audio objects are classified by a class category, by which they are can be assigned class specific processing. Audio objects classes can also have a rank. The rank of an audio objects class is used to give priority to or apply specific processing to audio objects in the presence of other audio objects of different classes.

Abstract: Methods and apparatuses for context assisted noise reduction are disclosed. In one example, noise data associated with background noise detected by a microphone at a mobile device is received. The noise data is processed to identify whether a threshold noise level has been exceeded. An event notification is transmitted, where the event notification is operable to initiate identifying a location having a reduced background noise.

Abstract: Systems and methods for automatic user specific, condition specific communication system intelligibility testing and optimization are provided. The intelligibility of speech for a particular user is determined using a test of intelligibility administered by an interactive voice response (IVR) application running on a communication server. The intelligibility test can be run for a particular user under different conditions. For each user and/or set of conditions, a set of speech signal adjustment parameters can be determined. A set of speech signal adjustment parameters that will enhance the intelligibility of a speech signal for a user are applied when that user is involved in a communication session. The particular set of speech signal adjustment parameters selected can depend on the communication equipment and/or environment associated with the communication session.

Abstract: In a selective signal encoder, an input signal is first encoded using a core layer encoder to produce a core layer encoded signal. The core layer encoded signal is decoded to produce a reconstructed signal and an error signal is generated as the difference between the reconstructed signal and the input signal. The reconstructed signal is compared to the input signal. One of two or more enhancement layer encoders selected dependent upon the comparison and used to encode the error signal. The core layer encoded signal, the enhancement layer encoded signal and the selection indicator are output to the channel (for transmission or storage, for example).

Abstract: Systems and methods are described that apply a noise feedback coding (NFC) technique at the encoder of a delta modulation codec, such as a Continuously Variable Slope Delta Modulation (CVSD) codec, so as to shape the spectrum of the coding noise produced thereby in such a way that the speech quality of the delta modulation decoder output is enhanced. The techniques described herein are not limited to delta modulation codecs and may also be applied to any sample-by-sample codec, including a G.711 ?-law codec, a linear pulse code modulation (LPCM), or any other of a wide variety of sample-by-sample codecs, to improve the audio quality of the decoder output thereof.

Abstract: The detection of double-talk in audio communication is provided. A communication device may receive an echo signal mixed with a speech signal at a near end location. The echo signal may be generated by speech transmitted by a remote party at a far end location to a local party at the near end location. The speech signal may be received from the local party for transmission to the remote party. The communication device may then filter the echo signal and the speech signal. The communication device may then analyze the speech signal to identify speech characteristics which indicate the presence of double-talk. The communication device may then set a flag upon identifying the speech characteristics which indicate the presence of the double-talk. The communication device may then process the filtered signals to further suppress remaining echo prior to transmission of the speech signal to the remote party.

Abstract: A system and method provides auxiliary voice input to a mobile communication device (MCD). The system comprises an electronic skin tattoo capable of being applied to a throat region of a body. The electronic skin tattoo can include an embedded microphone; a transceiver for enabling wireless communication with the MCD; and a power supply configured to receive energizing signals from a personal area network associated with the MCD. A controller is communicatively coupled to the power supply. The controller can be configured to receive a signal from the MCD to initiate reception of an audio stream picked up from the throat region of the body for subsequent audio detection by the MCD under an improved signal-to-noise ratio than without the employment of the electronic skin tattoo.

Abstract: A method is disclosed for maintaining spatial queues in digital sound signals. Sound signals are received from each of a plurality of transducers. The sound signals are transformed using a common real-valued spectral gain, G, to maintain spatial cues within the sound signals, the common spectral gain, G, determined by: calculating G as a function of a derivative of a known cost function and as a function of at least one multichannel frequency-domain Bayesian short-time estimator.

Abstract: Automatic correcting of user's speech impairment in speech may include obtaining the audio signal of a given user's speech, and analyzing the obtained audio signal to identify artifacts caused by the user's impairment. The obtained audio signal may be modified by eliminating the identified artifacts from it. The modified audio signal may be provided, e.g., to be played or broadcast or transmitted.

Abstract: Automatic correcting of user's speech impairment in speech may include obtaining the audio signal of a given user's speech, and analyzing the obtained audio signal to identify artifacts caused by the user's impairment. The obtained audio signal may be modified by eliminating the identified artifacts from it. The modified audio signal may be provided, e.g., to be played or broadcast or transmitted.

Abstract: Speech processing for a vehicle, including receiving speech from a user via at least one speech microphone that converts the speech into a speech signal, receiving vehicle noise via at least one active noise control microphone that converts the noise into a vehicle noise signal, and processing the speech signal in response to the vehicle noise signal to reduce vehicle noise in the speech signal.

Abstract: Enhanced speech is produced from a mixed signal including noise and the speech. The noise in the mixed signal is estimated using a vector-Taylor series. The estimated noise is in terms of a minimum mean-squared error. Then, the noise is subtracted from the mixed signal to obtain the enhanced speech.

Abstract: In speech processing systems, compensation is made for sudden changes in the background noise in the average signal-to-noise ratio (SNR) calculation. SNR outlier filtering may be used, alone or in conjunction with weighting the average SNR. Adaptive weights may be applied on the SNRs per band before computing the average SNR. The weighting function can be a function of noise level, noise type, and/or instantaneous SNR value. Another weighting mechanism applies a null filtering or outlier filtering which sets the weight in a particular band to be zero. This particular band may be characterized as the one that exhibits an SNR that is several times higher than the SNRs in other bands.

Abstract: A vehicle voice interface system calibration method comprising electronically convolving voice command data with voice impulse response data, electronically convolving audio system output data with feedback impulse response data, and calibrating the vehicle voice interface system. The voice command data is electronically convolved with voice impulse response data representing a voice acoustic signal path between an artificial mouth simulator and a first microphone, to simulate a voice acoustic transfer function pertaining to the passenger compartment. The audio system output data is convolved with feedback impulse response data representing a feedback acoustic signal path between a vehicle audio system output and a second microphone, to simulate a feedback acoustic transfer function pertaining to the passenger compartment. The voice interface system is calibrated to recognize voice commands represented by the voice command data based on the simulated voice and feedback acoustic transfer functions.

Abstract: A continuous stream of noise is created from a plurality of input signals. A smoothing spectrum estimate is continuously calculated from the continuous stream of noise. Noise is responsively removed from a selected one of the plurality of input signals using the smoothing spectrum estimate. The removal of the noise from the selected input signal is performed substantially synchronously and in time alignment with the creating of the continuous stream of noise and the calculating of the smoothing spectrum estimate.

Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for processing speech. A system configured to practice the method monitors user utterances to generate a conversation context. Then the system receives a current user utterance independent of non-natural language input intended to trigger speech processing. The system compares the current user utterance to the conversation context to generate a context similarity score, and if the context similarity score is above a threshold, incorporates the current user utterance into the conversation context. If the context similarity score is below the threshold, the system discards the current user utterance. The system can compare the current user utterance to the conversation context based on an n-gram distribution, a perplexity score, and a perplexity threshold. Alternately, the system can use a task model to compare the current user utterance to the conversation context.

Abstract: An electrical apparatus a voice signal receiving method thereof are disclosed. The electrical apparatus includes a plurality of voice receivers, a voice activity detector, a voice channel switch and a noise eliminator. The voice receivers are used to receive the voice signals. The voice activity detector receives and detects the voice signals, and obtains a main voice signal from the voice signals. The voice channel switch transports the main voice signal to a voice transporting channel and transports a plurality of other voice signals of the voice signals other than the main voice signal to a noise transporting channel according to a detecting result of the voice activity detector. The noise eliminator reduces the noise in the main voice according to the voice signals from the noise transporting channel.

Abstract: A method and apparatus are provided for generating a noise reduced output signal from sound received by a first microphone. The method includes transforming the sound received by the first microphone into a first input signal and transforming sound received by a second microphone into a second input signal. The method includes calculating, for each of a plurality of frequency components, an energy transfer function value as a real-valued quotient by dividing a temporally averaged product of an amplitude of the first input signal and the second input signal by a temporally averaged absolute square of the second input signal, calculating a gain value as a function of the calculated energy transfer function value, and generating the noise reduced output signal based on the product of the first input signal and the calculated gain value at each of the plurality of frequency components.

Abstract: An electronic device and a control method are provided. The electronic device includes a voice receiver which receives a voice of a user; a signal processor which performs signal processing on the received voice; a communicator which communicates with a first external device; and a controller which determines a text corresponding to the received voice of the user, and controls the communicator to transmit the signal processed voice and the determined text to the first external device.

Abstract: It is determined whether or not a sound picked up by at least either a first microphone or a second microphone is a speech segment. When it is determined that the sound picked up by the first or the second microphone is the speech segment, a voice incoming direction indicating from which direction a voice sound travels is detected based on a first sound pick-up signal obtained based on a sound picked up by the first microphone and a second sound pick-up signal obtained based on a sound picked up by the second microphone. A noise reduction process is performed using the first and second sound pick-up signals based on speech segment information indicating that the sound picked up by the first or the second microphone is the speech segment and voice incoming-direction information indicating the voice incoming direction.

Abstract: A voice signal processor detects background noise sections to reflect characteristics of the background noise on the Wiener filter coefficient to be used for suppressing noise components of input voice signals. In the voice signal processor, directivity signal generators form directivity signals having a directivity pattern. The directivity signals are used by a coherence calculator to obtain coherence, which is in turn used by a targeted voice section detector to detect a targeted voice section. A background noise section detector detects background noise sections containing no voice signal. When a background noise section is detected, a WF adapter uses characteristics of background noise in the detected temporal section to calculate a new WF coefficient.

Abstract: Provided are an apparatus and method for eliminating noise. The method includes: detecting a speech section from a noise speech signal including a noise signal; separating the speech section into a consonant section and a vowel section on the basis of a VOP at the speech section; calculating a transfer function of a filter for eliminating the noise signal to allow the degree of noise elimination to be different in the consonant section and the vowel section; and eliminating the noise signal from the noise speech signal on the basis of the transfer function.

Abstract: A method, system, and computer program product for managing noise in a noise reduction system, comprising: receiving a first signal at a first microphone; receiving a second signal at a second microphone; identifying noise estimation in the first signal and the second signal; identifying a transfer function of the noise reduction system using a ratio of a power spectral density of the second signal minus the noise estimation to a power spectral density of the first signal, wherein the noise estimation is removed from only the power spectral density of the second signal; and identifying a gain of the noise reduction system using the transfer function.

Abstract: A first signal is received that represents speech and the noise. The noise includes directional noise and diffused noise. A second signal is received that represents the noise and leakage of the speech. In response to the first and second signals: a first channel is generated that represents the speech and the diffused noise while attenuating most of the directional noise from the first signal; and a second channel is generated that represents the noise while attenuating most of the speech from the second signal. In response to the first and second channels, an output channel is generated that represents the speech while attenuating most of the noise from the first channel.

Abstract: At least one signal is received that represents speech and noise. In response to the at least one signal, frequency bands are generated of an output channel that represents the speech while attenuating at least some of the noise from the at least one signal. Within a kth frequency band of the at least one signal: a first ratio is determined of a clean version of the speech for a preceding time frame to the noise for the preceding time frame; and a second ratio is determined of a noisy version of the speech for the time frame n to the noise for the time frame n. In response to the first and second ratios, a gain is determined for the kth frequency band of the output channel for the time frame n.

Abstract: Methods and apparatus are provided for acoustic echo cancellation in a speech signal. Acoustic echo is cancelled by inserting at least one tone in the speech signal, wherein the at least one tone is substantially inaudible to a listener; determining a clock skew between two sampling clocks based on a frequency shift of the at least one tone; re-sampling the speech signal based on the determined clock skew; and performing the acoustic echo cancellation using the re-sampled speech signal. The provided acoustic echo cancellers can be implemented, for example, as terminal-based and/or network-based acoustic echo cancellers. The tone optionally comprises an inaudible tone or multiple tones. The tone generation can be limited to only when a speech power in the vicinity of the tone frequency is larger than a pre-determined threshold, or to the beginning of a call. A level of the tone can optionally be controlled so that the tone is masked by the speech signal.

Abstract: In response to a first envelope within a kth frequency band of a first channel, a speech level within the kth frequency band of the first channel is estimated. In response to a second envelope within the kth frequency band of a second channel, a noise level within the kth frequency band of the second channel is estimated. A noise suppression gain for a time frame n is computed in response to the estimated speech level for a preceding time frame, the estimated noise level for the preceding time frame, the estimated speech level for the time frame n, and the estimated noise level for the time frame n. An output channel is generated in response to multiplying the noise suppression gain for the time frame n and the first channel.

Abstract: Disclosed herein are systems, computer-implemented methods, and computer-readable storage media for tagging a known signal of interest. Initially, the system classifies the data from an input signal using a short-term classifier, wherein there are at least two classifications available, a first classification of the data as having no identified outputs and a second classification of the data as at least one potential signal of interest, wherein the short-term classifier also bypasses data that is known to be of no interest. After the short-term classifier classifies the inputs, it collapses the input data that is classified as having no identified outputs. This allows the short-term classifier to create time-variant data. Finally, the system will tag a known signal of interest in the time-variant data that was classified as having at least one potential signal of interest. Therefore, a system for tagging a known signal of interest is described.

Abstract: Described herein is a speech enhancement method using microphone arrays and a new iterative technique for enhancing noisy speech signals under low signal-to-noise-ratio (SNR) environments. Included is the processing of observed noisy speech both in the spatial- and the temporal-domains to enhance the desired signal component speech and an iterative technique to compute the generalized eigenvectors of the multichannel data derived from the microphone array. The entire processing is done on the spatio-temporal correlation coefficient sequence of the observed data in order to avoid large matrix-vector multiplications. Also described is a speech enhancement system having two stages. In the first stage, the noise component of the observed signal is whitened, and in the second stage a spatio-temporal power method is used to extract the most dominant speech component. In both the stages, the filters are adapted using the multichannel spatio-temporal correlation coefficients of the data.

Abstract: A system or method may facilitate delivery of network-specific dialing codes to a mobile node. When a mobile node is registered to a network part of the network infrastructure of a radio communication system, a request is generated by the mobile node, requesting download thereto of the dialing codes used in the network part to call service centers associated therewith. The requested dialing codes are downloaded to the mobile node. The downloaded dialing codes are indexed together with the dialing codes normally used by the mobile node to call the corresponding service centers. Subsequently, when a call is placed to a service center, the dialing codes are transposed, if necessary, to permit the call to a designated service center to be completed.

Abstract: The suppression of off-axis audio in an audio environment is provided. Off-axis audio may be considered audio that does not originate from a region of interest. The off-axis audio is suppressed by comparing a phase difference between signals from two microphones to a target slope of the phase difference between signals originating from the region of interest. The target slope can be adapted to allow the region of interest to move with the location of a human speaker such as a driver.

Abstract: The present invention discloses a speech enhancing method, a speech enhancing device and a denoising communication headphone.

Abstract: A method of processing audio signals during a communication session between a user device and a remote node, includes receiving a plurality of audio signals at audio input means at the user device including at least one primary audio signal and unwanted signals and receiving direction of arrival information of the audio signals at a noise suppression means. Known direction of arrival information representative of at least some of said unwanted signals is provided to the noise suppression means and the audio signals are processed at the noise suppression means to treat as noise, portions of the signal identified as unwanted dependent on a comparison between the direction of arrival information of the audio signals and the known direction of arrival information.

Abstract: A method and system for filtering a multi-channel audio signal having a speech channel and at least one non-speech channel, to improve intelligibility of speech determined by the signal. In typical embodiments, the method includes steps of determining at least one attenuation control value indicative of a measure of similarity between speech-related content determined by the speech channel and speech-related content determined by the non-speech channel, and attenuating the non-speech channel in response to the at least one attenuation control value. Typically, the attenuating step includes scaling of a raw attenuation control signal (e.g., a ducking gain control signal) for the non-speech channel in response to the at least one attenuation control value. Some embodiments are a general or special purpose processor programmed with software or firmware and/or otherwise configured to perform filtering in accordance the invention.

Abstract: A voice recognition device includes a voice recognition dictionary in which a word which is recognized as a result of voice recognition on an inputted voice is registered, a reply voice data storage unit for storing recorded voice data about words registered in the voice recognition dictionary, a dialog control unit for, when a word registered in the voice recognition dictionary is recognized, acquiring recorded voice data corresponding to the word from the reply voice data storage unit, a reproduction noise reduction unit for carrying out a process of reducing noise included in the recorded voice data, an amplitude adjusting unit for adjusting an amplitude of the recorded voice data in which the noise has been reduced to a predetermined amplitude level, and a voice reproduction unit for reproducing a voice from the amplitude-adjusted recorded voice data.

Abstract: A portable voice capture device comprising: an orientable arm with a first differential array of microphones comprising at least one pair of microphones, the directivity of said first array being arranged for sensing voice from a first direction depending on the orientation of said arm; a second differential array of microphones comprising at least one pair of microphones, the directivity of said second array being arranged for sensing noise from a second direction different from the first direction; a noise reduction circuit for providing a voice signal with reduced noise, based on the output of said first array and on the output of said second array.

Abstract: A space-time adaptive beamformer for reducing noise in a vehicle that includes two or more microphones. A first weighting network is used for adjusting the signal characteristics of at least one output of the microphones while at least one delay network is also used for delaying the output in time of at least one output of the microphones. A second weighting network then adjusts the signal characteristics of the output of each of the delay networks and at sum adder works to combine the output of the first weighting network and the second weighting network. Finally, an output of the sum adder is combined with an artificial noise free reference signal to provide a low distortion noise reduced output. By generating a desired signal that acts as an artificial noise free signal reference, adequate noise reduction to be obtained without the distortion created due to processing non-linearity.

Abstract: An electronic device for generating a masking signal is described. The electronic device includes a plurality of microphones and a speaker. The electronic device also includes a processor and executable instructions stored in memory that is in electronic communication with the processor. The electronic device obtains a plurality of audio signals from the plurality of microphones. The electronic device also obtains an ambience signal based on the plurality of audio signals. The electronic device further determines an ambience feature based on the ambience signal. Additionally, the electronic device obtains a voice signal based on the plurality of audio signals. The electronic device also determines a voice feature based on the voice signal. The electronic device additionally generates a masking signal based on the voice feature and the ambience feature. The electronic device further outputs the masking signal using the speaker.

Abstract: By monitoring the wind noise in a location in which a cellular telephone is operating and by applying noise reduction and/or cancellation protocols at the appropriate time via analog and/or digital signal processing, it is possible to significantly reduce wind noise entering into a communication system.

Abstract: The equipment comprises two microphones, sampling means, and de-noising means. The de-noising means are non-frequency noise reduction means comprising a combiner having an adaptive filter performing an iterative search seeking to cancel the noise picked up by one of the microphones on the basis of a noise reference given by the other microphone sensor. The adaptive filter is a fractional delay filter modeling a delay that is shorter than the sampling period. The equipment also has voice activity detector means delivering a signal representative of the presence or the absence of speech from the user of the equipment. The adaptive filter receives this signal as input so as to enable it to act selectively: i) either to perform an adaptive search for the parameters of the filter in the absence of speech; ii) or else to “freeze” those parameters of the filter in the presence of speech.

Abstract: An audio apparatus including a decorrelator for generating decorrelated signals by applying a phase shifting value adjusted based on a correlation difference between audio signals included in a multi-channel signal to the audio signals; and a speaker set including at least two speakers for outputting acoustic signals corresponding to the decorrelated signals

Abstract: In one implementation, a first voice stream for a packet-switched call is received from a calling party. The first voice stream conforms to a first silence suppression scheme and comprises a plurality of encoded packets for the packet-switched call. A subset of encoded packets are selected from the plurality of encoded packets to create a second voice stream that conforms to a second silence suppression scheme. The second voice stream comprises the subset of encoded packets. The first silence suppression scheme is distinct from the second silence suppression scheme. The second voice stream is forwarded toward a called party for the packet-switched call.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for enhancing speech recognition accuracy. In one aspect, a method includes receiving an audio signal that corresponds to an utterance recorded by a mobile device, determining a geographic location associated with the mobile device, identifying a set of geotagged audio signals that correspond to environmental audio associated with the geographic location, weighting each geotagged audio signal of the set of geotagged audio signals based on metadata associated with the respective geotagged audio signal, and using the set of weighted geotagged audio signals to perform noise compensation on the audio signal that corresponds to the utterance.

Abstract: A method, an apparatus, and a computer program, which can suppress a low frequency range component with a small amount of calculation, and can achieve a noise suppression of high quality, are provided. The noise superposed in a desired signal of an input signal is suppressed by converting the input signal to a frequency domain signal; correcting an amplitude of the frequency domain signal to obtain an amplitude corrected signal; obtaining an estimated noise by using the amplitude corrected signal; determining a suppression coefficient by using the estimated noise and the amplitude corrected signal; and weighting the amplitude corrected signal with the suppression coefficient.