Abstract: A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag. The high frequency regeneration is performed as a post-processing operation with a delay of 3010 samples per audio channel.

Abstract: Techniques are provided for acoustic event detection. A methodology implementing the techniques according to an embodiment includes extracting acoustic features from a received audio signal. The acoustic features may include, for example, one or more short-term Fourier transform frames, or other spectral energy characteristics, of the audio signal. The method also includes applying a trained classifier to the extracted acoustic features to identify and label acoustic event subparts of the audio signal and to generate scores associated with the subparts. The method further includes performing sequence decoding of the acoustic event subparts and associated scores to detect target acoustic events of interest based on the scores and temporal ordering sequence of the event subparts. The classifier is trained on acoustic event subparts that are generated through unsupervised subspace clustering techniques applied to training data that includes target acoustic events.

Abstract: Detecting fricatives in a noisy speech signal having a clean speech signal and a noise signal, includes bandpass filtering of the noisy speech signal with a first transfer function having a first passband range to provide a first filtered noisy speech signal, and bandpass filtering of the noisy speech signal with a second transfer function having a second passband range, the second passband being different from the first passband to provide a second filtered noisy speech signal. Detecting fricatives further includes applying a maximum operation to the first filtered noisy speech signal and the second filtered noisy speech signal to provide a maximum spectrum that is representative of a frequency range of maximum fricative energy, and deciding, based on the maximum spectrum, whether a fricative is contained in the noisy speech signal. A decision signal is output that is representative of the decision.

Abstract: One embodiment provides a method, including: identifying, using at least one sensor, that a position of a microphone attached to a headset is associated with an audible input position; determining, using a processor, that the position is not associated with an optimal audible input position; and notifying, responsive to the determining, a user that the position is not associated with the optimal audible input position. Other aspects are described and claimed.

Abstract: A head-worn audio device is provided with a circuit for voice signal enhancement. The circuit comprises at least a plurality of microphones, arranged at predefined positions, where each microphone provides a microphone signal. The circuit further comprises a directivity pre-processor and a blind source separation processor. The directivity pre-processor is connected with the plurality of microphones to receive the microphone signals and being configured to provide at least a voice signal and a noise signal. Directivity pre-processing increases the mutual independence of the signals provided to the blind source separation processor and thus improves processing by blind source separation. The blind source separation processor receives at least the voice signal and the noise signal, and is configured to conduct blind source separation on at least the voice signal and the noise signal to provide at least an enhanced voice signal with reduced noise components.

Abstract: A signal encoding method and device are disclosed. The method includes, when an encoding manner of a previous frame of a currently-input frame is a continuous encoding manner, predicting a comfort noise that is generated by a decoder according to the currently-input frame when the currently-input frame is encoded into an SID frame, determining an actual silence signal, determining a deviation degree between the comfort noise and the actual silence signal, determining an encoding manner of the currently-input frame according to the deviation degree, and encoding the currently-input frame according to the encoding manner of the currently-input frame. It is determined, according to the deviation degree between the comfort noise and the actual silence signal, that the encoding manner of the currently-input frame is the hangover frame encoding manner or the SID frame encoding manner, which can save communication bandwidth.

Abstract: Apparatus having corresponding methods comprise a microphone configured to produce audio; a mute control configured to select a microphone open selection or a microphone muted selection; a processor configured to identify the audio produced during the microphone open selection as primary audio, and to identify the audio produced during the microphone muted selection as secondary audio; and a transceiver configured to transmit the primary audio over a first link and the secondary audio over a second link different than the first link.

Abstract: An apparatus for generating a frequency enhancement signal has: a signal generator for generating an enhancement signal from a core signal, the enhancement signal having an enhancement frequency range not included in the core signal, wherein a current time portion of the enhancement signal or the core signal has subband signals for a plurality of subbands; a controller for calculating the same smoothing information for the plurality of subband signals of the enhancement frequency range or the core signal, and wherein the signal generator is configured for smoothing the plurality of subband signals of the enhancement frequency range or the core signal using the same smoothing information.

Abstract: Example embodiments disclosed herein relate to impulsive noise suppression. A method of impulsive noise suppression in an audio signal is disclosed. The method includes determining an impulsive noise related feature from a current frame of the audio signal. The method also includes detecting an impulsive noise in the current frame based on the impulsive noise related feature, and in response to detecting the impulsive noise in the current frame, applying a suppression gain to the current frame to suppress the impulsive noise. Corresponding system and computer program product of impulsive noise suppression in an audio signal are also disclosed.

Abstract: Processing digitized microphone signal data in order to detect wind noise. A first signal and a second signal are obtained from at least one microphone. The first and second signals reflect a common acoustic input, and are either temporally distinct or spatially distinct, or both. The first signal is processed to determine a first distribution of the samples of the first signal. The second signal is processed to determine a second distribution of the samples of the second signal. A difference between the first distribution and the second distribution is calculated. If the difference exceeds a detection threshold, an indication is output that wind noise is present.

Abstract: Processing digitized microphone signal data in order to detect wind noise. A first signal and a second signal are obtained from at least one microphone. The first and second signals reflect a common acoustic input, and are either temporally distinct or spatially distinct, or both. The first signal is processed to determine a first distribution of the samples of the first signal. The second signal is processed to determine a second distribution of the samples of the second signal. A difference between the first distribution and the second distribution is calculated. If the difference exceeds a detection threshold, an indication is output that wind noise is present.

Abstract: A device includes a receiver, a memory, and a processor. The receiver is configured to receive a remote voice profile. The memory is electrically coupled to the receiver. The memory is configured to store a local voice profile associated with a person. The processor is electrically coupled to the memory and the receiver. The processor is configured to determine that the remote voice profile is associated with the person based on speech content associated with the remote voice profile or an identifier associated with the remote voice profile. The processor is also configured to select the local voice profile for profile management based on the determination.

Abstract: The methods, apparatus, and systems described herein are designed to identify fraudulent callers. A voice print of a call is created and compared to known voice prints to determine if it matches one or more of the known voice prints. The methods include a pre-processing step to separate speech from non-speech, selecting a number of elements that affect the voice print the most, and/or computing an adjustment factor based on the scores of each received voice print against known voice prints.

Abstract: Method and system for use with audible signals that analyzes the signals into time-frequency frames over first and second time periods. Estimates of the noise and/or reverberation are derived from the frames. Gains are derived from the estimates and raised to a power to create modified gains. The modified gains are applied to the frames in the appropriate time periods. Modified audible signals are output after being processed by the modified gains.

Abstract: A system and method for processing sounds are provided. The sound processing system comprises a sound sensing unit including a plurality of microphones, each microphone providing a non-manipulated sound signal; a beam synthesizer including a plurality of filters, wherein each filter corresponds to at least one parameter for generating at least one sound beam; a sound analyzer connected to the sound sensing unit and to the beam synthesizer, wherein the sound analyzer is configured to generate at least one manipulated sound signal responsive to the plurality of filters and to the non-manipulated sound signals provided by at least two of the microphones.

Abstract: An object sound period detection apparatus includes a first calculating unit, a second calculating unit, a first detecting unit, and a second detecting unit. The first calculating unit calculates a first threshold every unit time. The second calculating unit calculates a second threshold every unit time. The first detecting unit compares first feature amount based on the input signal with the first threshold and detects the object sound period in the input signal. The second detecting unit compares second feature amount based on the input signal with the second threshold, detects the object sound period in the input signal, and outputs a detecting result. The first calculating unit calculates the first threshold based on a detecting result before unit time by the second detecting unit. The second calculating unit calculates the second threshold based on a detecting result in same unit time by the first detecting unit.

Abstract: A device includes a receiver, a memory, and a processor. The receiver is configured to receive a remote voice profile. The memory is electrically coupled to the receiver. The memory is configured to store a local voice profile associated with a person. The processor is electrically coupled to the memory and the receiver. The processor is configured to determine that the remote voice profile is associated with the person based on speech content associated with the remote voice profile or an identifier associated with the remote voice profile. The processor is also configured to select the local voice profile for profile management based on the determination.

Abstract: A system and method relate to receiving, by a processing device, a plurality of sound signals captured at a plurality of microphone sensors, wherein the plurality of sound signals are from a sound source, and wherein a number (M) of the plurality of microphone sensors is greater than three, determining a number (K) of layers for a multistage minimum variance distortionless response (MVDR) beamformer based on the number (M) of the plurality of microphone sensors, wherein the number (K) of layers is greater than one, and wherein each layer of the multistage MVDR beamformer comprises one or more mini-length MVDR beamformers, and executing the multistage MVDR beamformer to the plurality of sound signals to calculate an estimate of the sound source.

Abstract: A system and method for use with a voice-capable system, includes but is not limited to receiving a vocal input to the voice-capable system, receiving one or more instructions referential to the first speech output version of the vocal input, and creating a second speech output version of the vocal input representational of the first speech output version of the vocal input manipulated responsive to the one or more instructions.

Abstract: A Voice Activity Detection/Silence Suppression (VAD/SS) system is connected to a channel of a transmission pipe. The channel provides a pathway for the transmission of energy. A method for operating a VAD/SS system includes detecting the energy on the channel, and activating or suppressing activation of the VAD/SS system depending upon the nature of the energy detected on the channel.

Abstract: A minutes making assistance device according to the present invention includes: a sound processing unit that performs processing regarding a voice and determines whether or not speaking is started; an operation processing unit that performs processing regarding an operation and determines whether or not the operation is performed; a display processing unit that performs processing regarding a display; and a control unit that stores speaking start time and warning time in a memory when the sound processing unit determines that the speaking is started, performs warning processing when the current time becomes the warning time, and terminates the processing when the operation processing unit determines that the operation is performed before the warning time.

Abstract: A noise filler for providing a noise-filled spectral representation of an audio signal on the basis of an input spectral representation of the audio signal has a spectral region identifier configured to identify spectral regions of the input spectral representation spaced from non-zero spectral regions of the input spectral representation by at least one intermediate spectral region, to obtain identified spectral regions, and a noise inserter configured to selectively introduce noise into the identified spectral regions to obtain the noise-filled spectral representation of the audio signal. A noise filling parameter calculator for providing a noise filling parameter on the basis of a quantized spectral representation of an audio signal has a spectral region identifier, as mentioned above, and a noise value calculator configured to selectively consider quantization errors of the identified spectral regions for a calculation of the noise filling parameter.

Abstract: A Voice Activity Detection/Silence Suppression (VAD/SS) system is connected to a channel of a transmission pipe. The channel provides a pathway for the transmission of energy. A method for operating a VAD/SS system includes detecting the energy on the channel, and activating or suppressing activation of the VAD/SS system depending upon the nature of the energy detected on the channel.

Abstract: Methods and systems are provided for filtering sound. A position sensor determines positions of a plurality of occupants in a defined space. Multiple microphones receive sound and generate corresponding audio signals. A processor in communication with the microphones and the position sensor receives the positions of the occupants and the audio signals. The processor determines which of the occupants are engaging in speech and applies a temporal-spatial filter to the audio signals to generate a plurality of output signals corresponding respectively to each occupant of the defined space.

Abstract: Embodiments of the present invention relate to a voice detector receiving an input signal that is divided into sub-signals that represent a frequency sub-band. The voice detector calculates, for each sub-band, a signal-to-noise (SNR) value based on a corresponding sub-signal for each sub-band and a background signal for each sub-band. The voice detector also calculates a power SNR value for each sub-band, where at least one of the power SNR values is calculated based on a non-linear function. The voice detector forms a single value based on the calculated power SNR values and compares the single value and a given threshold value to make a voice activity decision presented on an output port.

Abstract: Embodiments of methods and devices for classifying background noise contained in an audio signal are disclosed. In one embodiment, the device includes a module for extracting from the audio signal a background noise signal, termed the noise signal. Also included is a second that calculates a first parameter, termed the temporal indicator. The temporal indicator relates to the temporal evolution of the noise signal. The second module also calculates a second parameter, termed the frequency indicator. The frequency indicator relates to the frequency spectrum of the noise signal. Finally, the device includes a third module that classifies the background noise by selecting, as a function of the calculated values of the temporal indicator and of the frequency indicator, a class of background noise from among a predefined set of classes of background noise.

Abstract: One or more audio signals are processed using a multi-stage (hierarchical) voice and/or signal activity detector (VAD/SAD). A first stage is capable of reducing the workload bandwidth by employing an inexpensive VAD/SAD processor. One or more subsequent stages may further process the audio signals from the first stage. Other implementations may include a first stage that also performs continuity preservation between last blocks of audio signal and the first blocks of audio after it is detected that relevant audio signals are resumed. In yet other implementations, the first stage may extract features from audio signals when they are presented in their coded domain, and possibly with little or no decoding of the audio signal.

Abstract: Methods and systems for extracting speech from such packet streams. The methods and systems analyze the encoded speech in a given packet stream, and automatically identify the actual speech coding scheme that was used to produce it. These techniques may be used, for example, in interception systems where the identity of the actual speech coding scheme is sometimes unavailable or inaccessible. For instance, the identity of the actual speech coding scheme may be sent in a separate signaling stream that is not intercepted. As another example, the identity of the actual speech coding scheme may be sent in the same packet stream as the encoded speech, but in encrypted form.

Abstract: A frame extracting means 71 extracts frames from sample data as voice data in which whether each frame is an active voice frame or a non-active voice frame is already known. A feature quantity calculating means 72 calculates multiple feature quantities of each of the frames. A feature quantity integrating means 73 calculates an integrated feature quantity of the multiple feature quantities. A judgment means 74 judges whether each of the frames is an active voice frame or a non-active voice frame. An erroneous feature quantity calculation value calculating means 75 obtains a first erroneous feature quantity calculation value and a second erroneous feature quantity calculation value by executing prescribed calculations. A weight updating means 76 updates weights used for weighting so that the rate between the first erroneous feature quantity calculation value and the second erroneous feature quantity calculation value approaches a prescribed value.

Abstract: A signal bandwidth extending apparatus including: a bandwidth extending section configured to extend a frequency bandwidth of a target signal, the target signal included in an input signal; a calculating section configured to calculate a degree of the target signal included in the input signal; and a controller configured to change a method of extending the frequency bandwidth by the bandwidth extending section according to a result of the calculating section.

Abstract: A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract: The invention automatically enables and disables noise reduction based on a noise threshold. This threshold can be pre-defined by a user for a particular machine or can be defined “on the fly” before/during a telephonic conversation. With this flexibility, the users can “by-pass” the noise reduction and preserve the voice quality which are usually altered/modified by noise reduction algorithms. The present invention provides a novel system and method for monitoring the audio signals, analyze selected audio signal components, compare the results of analysis with a threshold value, and enable or disable noise reduction capability of a communication device.

Abstract: A Voice Activity Detection/Silence Suppression (VAD/SS) system is connected to a channel of a transmission pipe. The channel provides a pathway for the transmission of energy. A method for operating a VAD/SS system includes detecting the energy on the channel, and activating or suppressing activation of the VAD/SS system depending upon the nature of the energy detected on the channel.

Abstract: Embodiments of the present invention provide an adaptive noise canceling system. The adaptive noise canceling system may be used in a handset to cancel background noise by generating an anti-noise signal. The adaptive noise canceling system may include first input to receive a first signal from a feedforward microphone; a second input to receive a second signal from an error microphone; a controller coupled to the inputs, the controller configured to adaptively generate an anti-noise signal according to the received signals, wherein the controller derives a profile of the anti-noise signal from the first signal and derives a magnitude of the anti-noise signal from both first and second signal; and an output to transmit the anti-noise signal to a speaker.

Abstract: A mute setting is automatically set based on a speech detection result for acoustic signals received by a device. A device detects the speech based on a variety of cues from acoustic signals received using one or more microphones. If speech is detected within one or more frames, a mute setting may be automatically turned off. If speech is not detected, a mute setting may be automatically turned on. A mute setting may remain on as long as speech is not detected within the received acoustic signals. A varying delay may be implemented to help avoid false detections. The delay may be utilized during a mute-on state, and gradually removed during a transition from a mute-on state to a mute-off state.

Abstract: The application describes a method and an apparatus to prevent clipping of an audio signal when protection against signal clipping by received audio metadata is not guaranteed. The method may be used to prevent clipping for the case of downmixing a multichannel signal to a stereo audio signal. According to the method, it is determined whether first gain values (4) based on received audio metadata are sufficient for protection against clipping of the audio signal. The audio metadata is embedded in a first audio stream (1). In case a first gain value (4) is not sufficient for protection, the respective first gain value (4) is replaced with a gain value sufficient for protection against clipping of the audio signal. Preferably, in case no metadata related to dynamic range control is present in the first audio stream (1), the method may add gain values sufficient for protection against signal clipping.

Abstract: A voice activity detection (VAD) apparatus configured to provide a voice activity detection decision for an input audio signal. The VAD apparatus includes a state detector and a voice activity calculator. The state detector is configured to determine, based on the input audio signal, a current working state of the VAD apparatus among at least two different working states. Each of the at least two different working states is associated with a corresponding working state parameter decision set which includes at least one voice activity detection parameter. The voice activity calculator is configured to calculate a voice activity detection parameter value for the at least one voice activity detection parameter of the working state parameter decision set associated with the current working state, and to provide the voice activity detection decision by comparing the calculated voice activity detection parameter value with a threshold.

Abstract: A method and apparatus to evaluate a quality of an audio signal, in which the number of effective channels is determined for each of a reference signal of a current frame and a test signal indicative of the reference signal that has passed through an audio codec, and an audio quality evaluation score of the current frame is calculated by evaluating an audio quality of the current frame based on the determined number of effective channels for each of the reference signal and the test signal by means of a predetermined evaluator.

Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.

Abstract: A speech sound detection apparatus receives an input audio signal (as a sound reception unit), and computes input power that indicates a magnitude of the sound represented by the audio signal (as an input power computation unit). The apparatus estimates a correction function that is a continuous function defining a relation between a certain frequency and a correction coefficient used to approximate the input power computed at that frequency to the reference power predetermined for that frequency (as a correction function estimation unit). The apparatus corrects the input power at every frequency, based upon the correction coefficient that is obtained in accordance with the relation defined by the estimated correction function (as an input power correcting unit). The apparatus further determines whether or not the sound represented by the received audio signal is speech sound, based upon the corrected input power (as a speech sound detection unit).

Abstract: Presented is a method for executing a command on a computing device. A computing device receives a first command and a second command, wherein the second command is, optionally, silent speech. The first command and the second command are combined to provide a final command to the computing device for execution.

Abstract: In one configuration, erasure of a significant frame of a sustained voiced segment is detected. An adaptive codebook gain value for the erased frame is calculated based on the preceding frame. If the calculated value is less than (alternatively, not greater than) a threshold value, a higher adaptive codebook gain value is used for the erased frame. The higher value may be derived from the calculated value or selected from among one or more predefined values.

Abstract: Audio content is converted to text using speech recognition software. The text is then associated with a distinct voice or a generic placeholder label if no distinction can be made. From the text and voice information, a word cloud is generated based on key words and key speakers. A visualization of the cloud displays as it is being created. Words grow in size in relation to their dominance. When it is determined that the predominant words or speakers have changed, the word cloud is complete. That word cloud continues to be displayed statically and a new word cloud display begins based upon a new set of predominant words or a new predominant speaker or set of speakers. This process may continue until the meeting is concluded. At the end of the meeting, the completed visualization may be saved to a storage device, sent to selected individuals, removed, or any combination of the preceding.

Abstract: This application relates to a voice activity detection (VAD) apparatus configured to provide a voice activity detection decision for an input audio signal. The VAD apparatus includes a state detector and a voice activity calculator. The state detector is configured to determine, based on the input audio signal, a current working state of the VAD apparatus among at least two different working states. Each of the at least two different working states is associated with a corresponding working state parameter decision set which includes at least one voice activity decision parameter. The voice activity calculator is configured to calculate a voice activity detection parameter value for the at least one voice activity decision parameter of the working state parameter decision set associated with the current working state, and to provide the voice activity detection decision by comparing the calculated voice activity detection parameter value with a threshold.

Abstract: A non-speech section detecting device generating a plurality of frames having a given time length on the basis of sound data obtained by sampling sound, and detecting a non-speech section having a frame not containing voice data based on speech uttered by a person, the device including: a calculating part calculating a bias of a spectrum obtained by converting sound data of each frame into components on a frequency axis; a judging part judging whether the bias is greater than or equal to a given threshold or alternatively smaller than or equal to a given threshold; a counting part counting the number of consecutive frames judged as having a bias greater than or equal to the threshold or alternatively smaller than or equal to the threshold; a count judging part judging whether the obtained number of consecutive frames is greater than or equal to a given value.

Abstract: A scheme to judge emphasized speech portions, wherein the judgment is executed by a statistical processing in terms of a set of speech parameters including a fundamental frequency, power and a temporal variation of a dynamic measure and/or their derivatives. The emphasized speech portions are used for clues to summarize an audio content or a video content with a speech.

Abstract: A mobile communication system comprises a mobile communication device provided with a touch screen; and a speech activity analyzer suitable to receive from said touch screen data indicative of pressure applied to an area of said touch screen, and of changes thereto with time.

Abstract: A Yule-Walker based, low-complexity voice activity detector (VAD) is disclosed. An input signal is typically noisy speech (i.e., corrupted with, for example, babble noise). In one embodiment, a first initialization stage of the VAD computes an occurrence of a silent period within the input signal and the AR parameters. The VAD could accordingly compute a tentative adaptive threshold and output hypothesis H1 (which means speech is present) during this stage. During the second initialization stage, the VAD generally builds a database of associated values and computes the adaptive threshold accordingly. The second initialization stage could also output tentative VAD decisions based on the tentative threshold computed in the first initialization stage. Finally, the VAD periodically retrains or updates AR parameters, threshold values and/or the database and outputs VAD decisions accordingly.

Abstract: The present invention provides a novel system and method for monitoring the audio signals, analyze selected audio signal components, compare the results of analysis with a threshold value, and enable or disable noise reduction capability of a communication device.