Abstract: Embodiments of a system for identifying audio content are described. During operation, the system receives a data stream from an electronic device via a communication network. Then, the system distorts a set of target patterns which are used to identify the audio content based on characteristics of the electronic device and/or the communication network. Next, the system identifies the audio content in the data stream based on the set of distorted target patterns.

Abstract: A noise suppressor for altering a speech signal is trained based on a speech recognition system. An objective function can be utilized to adjust parameters of the noise suppressor. The noise suppressor can be used to alter speech signals for the speech recognition system.

Abstract: A method is disclosed for applying a multi-state barge-in acoustic model in a spoken dialogue system. The method includes receiving an audio speech input from the user during the presentation of a prompt, accumulating the audio speech input from the user, applying a non-speech component having at least two one-state Hidden Markov Models (HMMs) to the audio speech input from the user, applying a speech component having at least five three-state HMMs to the audio speech input from the user, in which each of the five three-state HMMs represents a different phonetic category, determining whether the audio speech input is a barge-in-speech input from the user, and if the audio speech input is determined to be the barge-in-speech input from the user, terminating the presentation of the prompt.

Abstract: A speech enhancement system improves the perceptual quality of a processed voice signal. The system improves the perceptual quality of a voice signal by removing unwanted noise components from a voice signal. The system removes undesirable signals that may result in the loss of information. The system receives and analyzes signals to determine whether an undesired random or persistent signal corresponds to one or more modeled noises. When one or more noise components are detected, the noise components are substantially removed or dampened from the signal to provide a less noisy voice signal.

Abstract: A portable terminal having an audio pickup means that acquires sound, an absolute position detection unit that detects the absolute position of the portable terminal, a relative position detection unit that detects the relative position of the portable terminal, and a speech recognition and synthesis unit that recognizes the audio acquired by the audio pickup means as speech, is achieved with a simple configuration. A portable terminal (1) that exchanges data with a server (2) has disposed to the portable terminal an audio pickup means that acquires sound, an absolute position detection unit (1-1) that detects the absolute position of the portable terminal, a relative position detection unit (1-2) that detects the relative position of the portable terminal, and a speech recognition and synthesis unit (1-3) that recognizes the audio acquired by the audio pickup means as speech.

Abstract: Digital signal processing techniques for automatically reducing audible noise from a sound recording that contains speech. A noise suppression system uses two types of noise estimators, including a more aggressive one and less aggressive one. Decisions are made on how to select or combine their outputs into a usable noise estimate in a different speech and noise conditions. A 2-channel noise estimator is described. Other embodiments are also described and claimed.

Abstract: This invention presents a noise cancellation device for improved personal face-to-face and radio communications in high noise environments. The device comprises speech acquisition components, an audio signal processing module, a loudspeaker, and a radio interface. With the noise cancellation device, the signal-to-noise ratio can be improved by as much as 30 dB.

Abstract: The present technology provides noise reduction of an acoustic signal using a configurable classification threshold which provides a sophisticated level of control to balance the tradeoff between positional robustness and noise reduction robustness. The configurable classification threshold corresponds to a configurable spatial region, such that signals arising from sources within the configurable spatial region are preserved, and signals arising from sources outside it are rejected. In embodiments, the configurable classification threshold can be automatically and dynamically adjusted in real-time based on evaluated environmental conditions surrounding an audio device implementing the noise reduction techniques described herein.

Abstract: An imaging apparatus is provided. The apparatus includes a sound collecting unit configured to collect speech in a monitored environment, a shooting unit configured to shoot video in the monitored environment, a detection unit configured to detect a change in a state of the monitored environment based upon a change in data acquired by the sound collecting unit, the shooting unit and a sensor for measuring the state of the monitored environment, a recognition unit configured to recognize the change in state with regard to speech data acquired by the sound collecting unit and video data acquired by the shooting unit, and a control unit configured to start up the recognition unit and select a recognition database, which is used by the recognition unit, based upon result of detection by the detection unit.

Abstract: VoIP phones according to the present invention include a microphone, which may be internal or external, and allow the user to communicate unobtrusively, check voice mail and conduct other activities in an environment which can be noisy in general and extremely noisy sometimes. Speech recognition functionally may also be used to generate and send touch tone or DTMF tones such as in response to call trees or voice recognition functionality used by airlines, credit card companies, voice mail systems, and other applications. A system and method of audio processing which provides enhanced speech recognition is provided. Audio input is received at the microphone which is processed by adaptive noise cancellation to generate an enhanced audio signal. The operation of the speech recognition engine and the adaptive noise canceller may be advantageously controlled based on Voice Activity Detection (VAD).

Abstract: The present invention relates to means and methods of classifying speech and music signals in voice communication systems, devices, telephones, and methods, and more specifically, to systems, devices, and methods that automate control when either speech or music is detected over communication links. The present invention provides a novel system and method for monitoring the audio signal, analyze selected audio signal components, compare the results of analysis with a pre-determined threshold value, and classify the audio signal either as speech or music.

Abstract: A method of suppressing wind noise in a voice signal determines an upper frequency limit that lies within the frequency spectrum of the voice signal, and for each of a plurality of frequency bands below the upper frequency limit, compares the average power of signal components in a first portion of the signal to the average power of signal components in a second portion of the signal, where the second portion is successive to the first portion. Signal components are identified in at least one of the plurality of frequency bands as containing impulsive wind noise in dependence on the comparison, and the identified signal components are attenuated.

Abstract: An audio data processing apparatus and method to reduce wind noise. The apparatus includes a wind noise detecting unit to detect a wind noise section from an input audio signal, and a signal processing unit to reduce a low-frequency band signal of the input audio signal in the detected wind noise section. The apparatus determines whether wind is present and automatically reduces wind noise based on the determined result. Accordingly, the apparatus can effectively reduce wind noise.

Abstract: Various embodiments of methods and systems for providing informational updates, such as directional updates, to a user of a computing device are disclosed. The method and system may recognize an active conversation and monitor the conversation for a lull. During a detected lull, the method and system may deliver the informational updates. Certain embodiments may include receiving a voice update while the user is using other functionality of the computing device, such as a telecommunication function. Embodiments may also include scheduling an update window within which the update may be delivered. The update window may be a predetermined time window in which a voice update is broadcast, for example. Embodiments may also include alerting the user to a pending update.

Abstract: A noise reduction is provided for a hearing apparatus, with which both stationary and also non-stationary interference noises can be attenuated in an input signal. An output signal is in this way to convey a quite sound impression. A signal processing is provided, which effects a noise reduction on the basis of two different methods. Provision is made on the one hand for a noise reduction for stationary interference noises and on the other hand for a noise reduction for spatially oriented interference noises. A selection facility selects between the two noise reductions.

Abstract: Systems, methods, and devices for noise profile determination for a voice-related feature of an electronic device are provided. In one example, an electronic device capable of such noise profile determination may include a microphone and data processing circuitry. When a voice-related feature of the electronic device is not in use, the microphone may obtain ambient sounds. The data processing circuitry may determine a noise profile based at least in part on the obtained ambient sounds. The noise profile may enable the data processing circuitry to at least partially filter other ambient sounds obtained when the voice-related feature of the electronic device is in use.

Abstract: This invention maps possibly noisy digital input from any of a number of different hardware or software sources such as keyboards, automatic speech recognition systems, cell phones, smart phones or the web onto an interpretation consisting of an action and one or more physical objects, such as robots, machinery, vehicles, etc. or digital objects such as data files, tables and databases. Tables and lists of (i) homonyms and misrecognitions, (ii) thematic relation patterns, and (iii) lexicons are used to generate alternative forms of the input which are scored to determine the best interpretation of the noisy input. The actions may be executed internally or output to any device which contains a digital component such as, but not limited to, a computer, a robot, a cell phone, a smart phone or the web. This invention may be implemented on sequential and parallel compute engines and systems.

Abstract: A device may include a communication interface configured to receive audio signals associated with audible communications from a user; an output device; and logic. The logic may be configured to determine one or more audio qualities associated with the audio signals, map the one or more audio qualities to at least one value, generate audio-related information based on the mapping, and provide, via the output device during the audible communications, the audio-related information to the user.

Abstract: Audio encoding method and device comprising the transmission, in addition to the data representing a frequency-limited signal, of information relating to a temporal filter that can be applied to the entire broadened signal, both in its transmitted low-frequency part and in its reconstituted high-frequency part. The application of this filter allowing the reshaping the reconstituted high-frequency part and the correction of compression artifacts present in the transmitted low-frequency part. In this way, the application of the temporal filter, simple and inexpensive, to all or part of the reconstituted signal, makes it possible to obtain a signal of good perceived quality.

Abstract: An operation control apparatus and method of controlling a plurality of operationally connected voice recognition-enabled systems, each having reciprocal control operational states corresponding to an enabled/disabled state.

Abstract: A speech recognition method and system, includes receiving in a first noise environment a speech input having a sequence of observations; determining a likelihood of a sequence of words arising from the sequence of observations using an acoustic model trained to recognize speech in a second noise environment, the model having a plurality of model parameters relating to the probability distribution of a word or part thereof being related to an observation; and adapting the model trained in the second environment to that of the first environment.

Abstract: Methods and apparatuses for performing song detection on an audio signal are described. Clips of the audio signal are classified into classes comprising music. Class boundaries of music clips are detected as candidate boundaries of a first type. Combinations including non-overlapped sections are derived. Each section meets the following conditions: 1) including at least one music segment longer than a predetermined minimum song duration, 2) shorter than a predetermined maximum song duration, 3) both starting and ending with a music clip, and 4) a proportion of the music clips in each of the sections is greater than a predetermined minimum proportion. In this way, various possible song partitions in the audio signal can be obtained for investigation.

Abstract: Audible navigation system produces direction updates, scheduled at predetermined time windows, during which the audio environment is monitored for the existence of a conversation. If no conversations are detected during an update window, or lulls in conversation are detected, the audible navigation system direction update is output. If uninterrupted conversations continue as the update window time is expiring, a system volume is lowered and the navigation system direction update is output.

Abstract: A wireless headset capable of receiving audio signals transmitted wirelessly and compatible for use in an MRI scanner is disclosed. The headset includes a first wireless module connected to the first earphone and a second wireless module connected to the second earphone. Each wireless module is electrically connected to a speaker in the respective earphone. The first wireless module receives the audio signal from a remote source and coordinates transmission of the audio signal to each of the speakers. The compact nature of each earphone minimizes the length of wire runs. In addition, the headset is made of materials having low magnetic susceptibility such that they will not be affected by the magnetic field from the MRI scanner.

Abstract: A continuous comfort noise is provided that is overlaid for the entire duration of a conference call scenario. The comfort noise may be adapted to match the levels of the actual background noise detected on one or more of the conference call participant's devices on the transmitting end(s) of a conference call as well as the participants' speech levels. The comfort noise may also be adapted to the type of listening device employed on the receiving end of a conference call. The comfort noise level may be customized to an appropriate and comfortable level for the type of listening device being used, and the system may continuously mix the comfort noise with incoming audio signals for the entire duration of a conference call, lowering the comfort noise level gradually during speaking periods for additional user experience improvement.

Abstract: To this end, a voice detection device includes a band-based power calculation unit that calculates a total of signal power values (sub-band power) of signals entered from the microphones from one preset frequency width (sub-band) to another. The voice detection device also includes a band-based noise estimation unit that estimates the sub-band based noise power, and a sub-band based SNR calculation unit. The sub-band based SNR calculation unit calculates a sub-band SNR from one sub-band to another to output the largest one of the sub-band SNRs as an SNR for a microphone of interest. The voice detection device further includes a voice/non-voice decision unit that determines the voice/non-voice using the SNR for the microphone of interest.

Abstract: Systems and methods are described for performing packet loss concealment (PLC) to mitigate the effect of one or more lost frames within a series of frames that represent a speech signal. In accordance with the exemplary systems and methods, PLC is performed by searching a codebook of speech-related parameter profiles to identify content that is being spoken and by selecting a profile associated with the identified content for use in predicting or estimating speech-related parameter information associated with one or more lost frames of a speech signal. The predicted/estimated speech-related parameter information is then used to synthesize one or more frames to replace the lost frame(s) of the speech signal.

Abstract: The disclosure provides a method and an apparatus for adaptively detecting a voice activity in an input audio signal composed of frames. The method comprises the steps of: determining a noise characteristic of the input signal based on a received frame of the input audio signal; deriving a voice activity detection (VAD) parameter based on the noise characteristic of the input audio signal; and comparing the derived VAD parameter with a threshold value to provide a voice activity detection decision.

Abstract: An embodiment of the invention provides a noise cancellation method for an electronic device. The method comprises: receiving an audio signal; applying a Fast Fourier Transform operation on the audio signal to generate a sound spectrum; acquiring a first spectrum corresponding to a noise and a second spectrum corresponding to a human voice signal from the sound spectrum; estimating a center frequency according to the first spectrum and the second spectrum; and applying a high pass filtering operation to the sound spectrum according to the center frequency.

Abstract: Described is a multiple phase process/system that combines spatial filtering with regularization to separate sound from different sources such as the speech of two different speakers. In a first phase, frequency domain signals corresponding to the sensed sounds are processed into separated spatially filtered signals including by inputting the signals into a plurality of beamformers (which may include nullformers) followed by nonlinear spatial filters. In a regularization phase, the separated spatially filtered signals are input into an independent component analysis mechanism that is configured with multi-tap filters, followed by secondary nonlinear spatial filters. Separated audio signals are the provided via an inverse-transform.

Abstract: An adaptive equalization system that adjusts the spectral shape of a speech signal based on an intelligibility measurement of the speech signal may improve the intelligibility of the output speech signal. Such an adaptive equalization system may include a speech intelligibility measurement module, a spectral shape adjustment module, and an adaptive equalization module. The speech intelligibility measurement module is configured to calculate a speech intelligibility measurement of a speech signal. The spectral shape adjustment module is configured to generate a weighted long-term speech curve based on a first predetermined long-term average speech curve, a second predetermined long-term average speech curve, and the speech intelligibility measurement. The adaptive equalization module is configured to adapt equalization coefficients for the speech signal based on the weighted long-term speech curve.

Abstract: A non-spatial speech detection system includes a plurality of microphones whose output is supplied to a fixed beamformer. An adaptive beamformer is used for receiving the output of the plurality of microphones and one or more processors are used for processing an output from the fixed beamformer and identifying speech from noise though the use of an algorithm utilizing a covariance matrix.

Abstract: A speech recognition system according to the present invention includes a sound source separating section which separates mixed speeches from multiple sound sources from one another; a mask generating section which generates a soft mask which can take continuous values between 0 and 1 for each frequency spectral component of a separated speech signal using distributions of speech signal and noise against separation reliability of the separated speech signal; and a speech recognizing section which recognizes speeches separated by the sound source separating section using soft masks generated by the mask generating section.

Abstract: An earpiece (100) and a method (300) personalized voice operable control can include capturing (302) an ambient sound from an Ambient Sound Microphone (111) to produce an electronic ambient signal (426), delivering (304) audio content (402) to an ear canal (131) by way of an Ear Canal Receiver (125) to produce an acoustic audio content (404) and capturing (306) in the ear canal an internal sound (402) from an Ear Canal Microphone (123) to produce an electronic internal signal (410). The electronic internal signal includes an echo of the acoustic audio content and a spoken voice generated by a wearer of the earpiece. The Method also includes detecting (312) the spoken voice in the electronic internal signal in the presence of the echo, and controlling (314) a voice operation of the earpiece when the spoken voice is detected.

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 recording medium is provided that records a separating step of separating a mixed sound signal in which a plurality of excitations are mixed into the respective excitations, and a step of performing speech detection on the plurality of separated excitation signals, judging whether or not the plurality of excitation signals are speech and generating speech section information indicating speech/non-speech information for each excitation signal. The recording medium also includes at least one of a step of calculating and analyzing an utterance overlap duration using the speech section information for combinations of the plurality of excitation signals and a step of calculating and analyzing a silence duration. The recording medium further includes a step of calculating a degree of establishment of a conversation indicating the degree of establishment of a conversation based on the extracted utterance overlap duration or the silence duration.

Abstract: A system and method are described for recognizing repeated audio material within at least one media stream without prior knowledge of the nature of the repeated material. The system and method are able to create a screening database from the media stream or streams. An unknown sample audio fragment is taken from the media stream and compared against the screening database to find if there are matching fragments within the media streams by determining if the unknown sample matches any samples in the screening database.

Abstract: An automatic speech recognition (ASR) system and method is provided for controlling the recognition of speech utterances generated by an end user operating a communications device. The ASR system and method can be used with a mobile device that is used in a communications network. The ASR system can be used for ASR of speech utterances input into a mobile device, to perform compensating techniques using at least one characteristic and for updating an ASR speech recognizer associated with the ASR system by determined and using a background noise value and a distortion value that is based on the features of the mobile device. The ASR system can be used to augment a limited data input capability of a mobile device, for example, caused by limited input devices physically located on the mobile device.

Abstract: An electronic device for suppressing noise in an audio signal is described. The electronic device includes a processor and instructions stored in memory. The electronic device receives an input audio signal and computes an overall noise estimate based on a stationary noise estimate, a non-stationary noise estimate and an excess noise estimate. The electronic device also computes an adaptive factor based on an input Signal-to-Noise Ratio (SNR) and one or more SNR limits. A set of gains is also computed using a spectral expansion gain function. The spectral expansion gain function is based on the overall noise estimate and the adaptive factor. The electronic device also applies the set of gains to the input audio signal to produce a noise-suppressed audio signal and provides the noise-suppressed audio signal.

Abstract: The invention relates to the processing of a digital signal originating from a decoder and a noise reduction post-processing step, including, in particular, limitation of distortion introduced by the post-processing step in order to deliver a corrected output signal (SOUT), assigning said corrected output signal (SOUT) with: a current amplitude having an intermediary value between a current amplitude value of the post-processed signal (SPOST) and a corresponding current amplitude value of the decoded signal (S?MIC), or the current amplitude of the post-processed signal (SPOST), according to the respective values of the current amplitude of the post-processed signal (SPOST) and by the corresponding current amplitude of the decoded signal (S?MIC).

Abstract: A method for restoring a processed speech signal by an electronic device is described. The method includes obtaining at least one audio signal. The method also includes performing bin-wise voice activity detection based on the at least one audio signal. The method further includes restoring the processed speech signal based on the bin-wise voice activity detection.

Abstract: A method for detecting voice activity by an electronic device is described. The method includes detecting near end speech based on a near end voiced speech detector and at least one single channel voice activity detector. The near end voiced speech detector is associated with a harmonic statistic based on a speech pitch histogram.

Abstract: A method and system for enhancing the frequency response of speech signals are provided. An average speech spectral shape estimate is calculated over time based on the input speech signal. The average speech spectral shape estimate may be calculated in the frequency domain using a first order IIR filtering or “leaky integrators.” Thus, the average speech spectral shape estimate adapts over time to changes in the acoustic characteristics of the voice path or any changes in the electrical audio path that may affect the frequency response of the system. A spectral correction factor may be determined by comparing the average speech spectral shape estimate to a desired target spectral shape. The spectral correction factor may be added (in units of dB) to the spectrum of the input speech signal in order to enhance or adjust the spectrum of the input speech signal toward the desired spectral shape, and an enhanced speech signal re-synthesized from the corrected spectrum.

Abstract: Methods related to Generalized Mutual Interdependence Analysis (GMIA), a low complexity statistical method for projecting data in a subspace that captures invariant properties of the data, are implemented on a processor based system. GMIA methods are applied to the signal processing problem of voice activity detection and classification. Real-world conversational speech data are modeled to fit the GMIA assumptions. Low complexity GMIA computations extract reliable features for classification of sound under noisy conditions and operate with small amounts of data. A speaker is characterized by a slow varying or invariant channel that is learned and is tracked from single channel data by GMIA methods.

Abstract: Speech enhancement based on a psycho-acoustic model is disclosed that is capable of preserving the fidelity of speech while sufficiently suppressing noise including the processing artifact known as “musical noise”.

Abstract: A decoding device is capable of flexibly calculating high-band spectrum data with a high accuracy in accordance with an encoding band selected by an upper-node layer of the encoding side. In this device: a first layer decoder decodes first layer encoded information to generate a first layer decoded signal; a second layer decoder decodes second layer encoded information to generate a second layer decoded signal; a spectrum decoder performs a band extension process by using the second layer decoded signal and the first layer decoded signal up-sampled in an up-sampler so as to generate an all-band decoded signal; and a switch outputs the first layer decoded signal or the all-band decoded signal according to the control information generated in a controller.

Abstract: A method of and system for transient noise rejection for improved speech recognition. The method comprises the steps of (a) receiving audio including user speech and at least some transient noise associated with the speech, (b) converting the received audio into digital data, (c) segmenting the digital data into acoustic frames, and (d) extracting acoustic feature vectors from the acoustic frames. The method also comprises the steps of (e) evaluating the acoustic frames for transient noise on a frame-by-frame basis, (f) rejecting those acoustic frames having transient noise, (g) accepting as speech frames those acoustic frames having no transient noise and, thereafter, (h) recognizing the user speech using the speech frames.

Abstract: A mobile terminal including an input unit configured to receive an input to activate a voice recognition function on the mobile terminal, a memory configured to store information related to operations performed on the mobile terminal, and a controller configured to activate the voice recognition function upon receiving the input to activate the voice recognition function, to determine a meaning of an input voice instruction based on at least one prior operation performed on the mobile terminal and a language included in the voice instruction, and to provide operations related to the determined meaning of the input voice instruction based on the at least one prior operation performed on the mobile terminal and the language included in the voice instruction and based on a probability that the determined meaning of the input voice instruction matches the information related to the operations of the mobile terminal.

Abstract: A method and apparatus for performing speech quality assessment in a speech communication system (such as, for example, a VoIP communication system) which detects and measures the presence of impulsive noise is provided. Specifically, in one illustrative embodiment, an autoregressive (AR) model of speech (and, in particular, of the excitation of the vocal tract) is advantageously employed to estimate a short-term variance of the speech excitation, and the standard deviation of the speech excitation (i.e., the square root of the variance) is then advantageously compared to a predetermined threshold to identify whether impulsive noise is present. Then, based on a statistic analysis of any such identified impulsive noise, a speech quality assessment is generated.

Abstract: A rule-based end-pointer isolates spoken utterances contained within an audio stream from background noise and non-speech transients. The rule-based end-pointer includes a plurality of rules to determine the beginning and/or end of a spoken utterance based on various speech characteristics. The rules may analyze an audio stream or a portion of an audio stream based upon an event, a combination of events, the duration of an event, or a duration relative to an event. The rules may be manually or dynamically customized depending upon factors that may include characteristics of the audio stream itself, an expected response contained within the audio stream, or environmental conditions.