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704 - Data processing: speech signal processing, linguistics, language translation, and audio compression/decompression

704200000 - SPEECH SIGNAL PROCESSING

704201000 - For storage or transmission

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Class / Patent application numberDescriptionNumber of patent applications / Date published
704228000 Post-transmission 17
704227000 Pretransmission 10
Entries
DocumentTitleDate
20130030801Off-Axis Audio Suppressions in An Automobile Cabin - 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.01-31-2013
20130211831SEMICONDUCTOR DEVICE AND VOICE COMMUNICATION DEVICE - A semiconductor device for realizing higher-precision noise elimination includes: a decoder which decodes an encoded input signal; a determining unit which determines whether or not a voice signal is included in the input signal; a suppressor which performs a suppressing process for suppressing a noise component included in the input signal on the basis of a result of determination by the determining unit; and a first storage for storing, as a determination criterion value used for the determination, a first criterion value which specifies the proportion of a voice signal with respect to voice distortion noise.08-15-2013
20130211830WIND SUPPRESSION/REPLACEMENT COMPONENT FOR USE WITH ELECTRONIC SYSTEMS - Techniques associated with an acoustic vibration sensor are described, including a first detector that receives a first signal and a second detector that receives a second signal and a third signal, wherein the first signal comprises a skin surface microphone signal, a static equalization filter coupled to the first detector and configured to generate an equalized first signal, a voice activity detector coupled to the first detector, and a wind detector coupled to the second detector, the wind detector configured to correlate the second signal and the third signal and to derive from the correlation a plurality of wind metrics associated with a wind noise, the wind detector is further configured to determine a magnitude associated with the wind noise, to determine whether to suspend an activity of the system, and to determine a duration of time that the magnitude associated with the wind noise exceeds a threshold.08-15-2013
20130211829ADAPTIVE SYSTEMS USING CORRENTROPY - Various methods and systems are provided for related to adaptive systems using correntropy. In one embodiment, a signal processing device includes a processing unit and a memory storing an adaptive system executable in the at least one processing unit. The adaptive system includes modules that, when executed by the processing unit, cause the signal processing device to adaptively filter a desired signal using a correntropy cost function. In another embodiment, a method includes adjusting a coefficient of an adaptive filter based at least in part on a correntropy cost function signal, providing an adaptive filter output signal based at least in part on the adjusted coefficient and a reference signal, and determining an error signal based at least in part on a received signal and the adaptive filter output signal.08-15-2013
20130211828SPEECH PROCESSING RESPONSIVE TO ACTIVE NOISE CONTROL MICROPHONES - 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.08-15-2013
20100076756SPATIO-TEMPORAL SPEECH ENHANCEMENT TECHNIQUE BASED ON GENERALIZED EIGENVALUE DECOMPOSITION - The present invention describes 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. A first embodiment involves the processing of the 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. A further embodiment relates to a speech enhancement system that is composed of 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 and hence avoid large matrix vector multiplications.03-25-2010
20130080158Speech Enhancement with Minimum Gating - A speech enhancement system enhances transitions between speech and non-speech segments. The system includes a background noise estimator that approximates the magnitude of a background noise of an input signal that includes a speech and a non-speech segment. A slave processor is programmed to perform the specialized task of modifying a spectral tilt of the input signal to match a plurality of expected spectral shapes selected by a Codec.03-28-2013
20100036659Noise-Reduction Processing of Speech Signals - The present invention relates to a method for signal processing comprising the steps of providing a set of prototype spectral envelopes, providing a set of reference noise prototypes, wherein the reference noise prototypes are obtained from at least a sub-set of the provided set of prototype spectral envelopes, detecting a verbal utterance by at least one microphone to obtain a microphone signal, processing the microphone signal for noise reduction based on the provided reference noise prototypes to obtain an enhanced signal and encoding the enhanced signal based on the provided prototype spectral envelopes to obtain an encoded enhanced signal.02-11-2010
20130035934DYNAMIC CONTROLLER FOR IMPROVING SPEECH INTELLIGIBILITY - 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.02-07-2013
20130041659SPATIO-TEMPORAL SPEECH ENHANCEMENT TECHNIQUE BASED ON GENERALIZED EIGENVALUE DECOMPOSITION - 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.02-14-2013
20130041660SYSTEM AND METHOD FOR TAGGING SIGNALS OF INTEREST IN TIME VARIANT DATA - 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.02-14-2013
20090157398Method and apparatus for detecting noise - A method of and apparatus for detecting noise are provided. The method of detecting noise includes: receiving an input of a voice frame and converting the voice frame into a filter bank vector; converting the converted filter bank vector into band data; calculating a weight Gaussian mixture model (GMM) for each band by using the converted band data; and detecting noise in the voice frame based on the calculation result.06-18-2009
20120185246NOISE SUPPRESSION USING MULTIPLE SENSORS OF A COMMUNICATION DEVICE - Techniques are described herein that suppress noise using multiple sensors (e.g., microphones) of a communication device. Noise modeling (e.g., estimation of noise basis vectors and noise weighting vectors) is performed with respect to a noise signal during operation of a communication device to provide a noise model. The noise model includes noise basis vectors and noise coefficients that represent noise provided by audio sources other than a user of the communication device. Speech modeling (e.g., estimation of speech basis vectors and speech weighting) is performed to provide a speech model. The speech model includes speech basis vectors and speech coefficients that represent speech of the user. A noisy speech signal is processed using the noise basis vectors, the noise coefficients, the speech basis vectors, and the speech coefficients to provide a clean speech signal.07-19-2012
20130046535Method, System and Computer Program Product for Suppressing Noise Using Multiple Signals - 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.02-21-2013
20100138220COMPUTER-READABLE MEDIUM FOR RECORDING AUDIO SIGNAL PROCESSING ESTIMATING PROGRAM AND AUDIO SIGNAL PROCESSING ESTIMATING DEVICE - A computer-readable medium recording a program allowing a computer to execute: setting a plurality of frames on a common time axis between a first waveform of an input to the audio processing and a second waveform of an output from the audio processing, detecting a voice frame and a noise frame in the first and second waveform, calculating a first and second spectrum from the first and second waveform, adjusting the level of the first or second spectrum of the noise frame, and setting the adjusted first and second spectrum of the noise frame as a third and fourth spectrum, calculating a distortion amount of the noise frame from the third and fourth spectrum, estimating a noise model spectrum from the first or second spectrum, and calculating a distortion amount of the voice frame from the first and second spectrum of the voice frame at the selected frequency.06-03-2010
20120191447METHOD AND APPARATUS FOR MASKING WIND NOISE - Wind and other noise is suppressed in a signal by adaptively changing characteristics of a filter. The filter characteristics are changed in response to the noise content of the signal over time using a history of noise content. Filter characteristics are changed according to a plurality of reference filters, the characteristics of which are chosen to optimally attenuate or amplify signals in a range of frequencies.07-26-2012
20130073284Speech Enhancement System - A speech enhancement system improves speech conversion within an encoder and decoder. The system includes a first device that converts sound waves into operational signals. A second device selects a template that represents an expected signal model. The selected template models speech characteristics of the operational signals through a speech codebook that is further accessed in a communication channel.03-21-2013
20130073283NOISE REDUCTION APPARATUS, AUDIO INPUT APPARATUS, WIRELESS COMMUNICATION APPARATUS, AND NOISE REDUCTION METHOD - 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.03-21-2013
20110015923METHOD AND APPARATUS FOR GENERATING NOISES - A method and an apparatus for generating comfortable noises so as to improve user experience are disclosed. The method includes: if a received data frame is a noise frame, calculating a corresponding energy attenuation parameter based on the noise frame and a data frame received earlier than the noise frame; and attenuating noise energy based on the energy attenuation parameter to obtain a comfortable noise signal. An apparatus for generating comfortable noise is also provided.01-20-2011
20120310637AUDIO EQUIPMENT INCLUDING MEANS FOR DE-NOISING A SPEECH SIGNAL BY FRACTIONAL DELAY FILTERING, IN PARTICULAR FOR A "HANDS-FREE" TELEPHONY SYSTEM - 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.12-06-2012
20090299740METHOD OF PROCESSING AUDIO SIGNALS FOR IMPROVING THE QUALITY OF OUTPUT AUDIO SIGNAL WHICH IS TRANSFERRED TO SUBSCRIBER'S TERMINAL OVER NETWORK AND AUDIO SIGNAL PRE-PROCESSING APPARATUS OF ENABLING THE METHOD - A method of pre-processing an audio signal transmitted to a user terminal via a communication network and an apparatus using the method are provided. The method of pre-processing the audio signal may prevent deterioration of a sound quality of the audio signal transmitted to the user terminal by pre-processing the audio signal, and by enabling a codec module, encoding the audio signal, to determine the audio signal as a speech signal. Also, the method of pre-processing the audio signal may improve a probability that the codec module may determine a corresponding audio signal as a speech when the audio signal is transmitted via the communication network by pre-processing the audio signal using a speech codec.12-03-2009
20090281803DISPERSION FILTERING FOR SPEECH INTELLIGIBILITY ENHANCEMENT - A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.11-12-2009
20090271187TWO MICROPHONE NOISE REDUCTION SYSTEM - A two microphone noise reduction system is described. In an embodiment, input signals from each of the microphones are divided into subbands and each subband is then filtered independently to separate noise and desired signals and to suppress non-stationary and stationary noise. Filtering methods used include adaptive decorrelation filtering. A post-processing module using adaptive noise cancellation like filtering algorithms may be used to further suppress stationary and non-stationary noise in the output signals from the adaptive decorrelation filtering and a single microphone noise reduction algorithm may be used to further provide optimal stationary noise reduction performance of the system.10-29-2009
20130066628APPARATUS AND METHOD FOR SUPPRESSING NOISE FROM VOICE SIGNAL BY ADAPTIVELY UPDATING WIENER FILTER COEFFICIENT BY MEANS OF COHERENCE - 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.03-14-2013
20090012783System and method for adaptive intelligent noise suppression - Systems and methods for adaptive intelligent noise suppression are provided. In exemplary embodiments, a primary acoustic signal is received. A speech distortion estimate is then determined based on the primary acoustic signal. The speech distortion estimate is used to derive control signals which adjust an enhancement filter. The enhancement filter is used to generate a plurality of gain masks, which may be applied to the primary acoustic signal to generate a noise suppressed signal.01-08-2009
20130166289VOICE EMPHASIS DEVICE - There is provided a voice emphasis device with which voice clarity can be improved. This voice emphasis device comprises a correlation component removal filter circuit that removes a correlation component from a voice signal produced at a specific sampling frequency, a multiplication circuit that produces an extracted signal by multiplying a specific gain coefficient by the output of the correlation component removal filter circuit, and an arithmetic circuit that adds or subtracts the extracted signal to or from the voice signal. The correlation component removal filter circuit is a lattice-type filter circuit that combines a feedforward filter and a feedback filter. The feedforward filter and the feedback filter update the filter coefficient at the specific sampling frequency based on the formula k06-27-2013
20080294430Noise reduction device, program and method - A noise reduction device is configured by use of: means for calculating a predetermined constant, and a predetermined reference signal Rω(T) in the frequency domain, respectively by use of adaptive coefficients Wω(m), and for thereby obtaining estimated values Nω and Qω(T) respectively of stationary noise components, and non-stationary noise components corresponding to the reference signal, which are included in a predetermined observed signal Xω(T) in the frequency domain; means and for applying a noise reduction process to the observed signal on the basis of each of the estimated values, and for updating each of the adaptive coefficients on the basis of a result of the process; and an adaptive learning means and for repeating the obtaining of the estimated values and the updating of the adaptive coefficients, and for thereby learning each of the adaptive coefficients.11-27-2008
20110282660System for Suppressing Rain Noise - A voice enhancement logic improves the perceptual quality of a processed signal. The voice enhancement system includes a noise detector and a noise attenuator. The noise detector detects and models the noise associated with rain. The noise attenuator dampens or reduces the rain noise from a signal to improve the intelligibility of an unvoiced, a fully voiced, or a mixed voice segment.11-17-2011
20110282659APPARATUS AND METHOD FOR IMPROVING COMMUNICATION SOUND QUALITY IN MOBILE TERMINAL - An apparatus and a method for improving communication sound quality in a mobile terminal in order to remove a neighboring noise that occurs together with a user's voice signal in a mobile terminal by discriminating signals occurring at different distances using two microphones and removing a noise. The mobile terminal preferably includes a first microphone, a second microphone, and a voice processor. The first microphone receives a voice signal occurring at a closer distance from the mobile terminal and a voice signal occurring at a longer distance from the mobile terminal. The second microphone receives only a voice signal occurring at the long distance. The voice processor discriminates between the signal occurring at the long distance and the signal occurring at the close distance by receiving voice signals received via the first microphone and the second microphone at different phases.11-17-2011
20110046947System and Method for Enhancing a Decoded Tonal Sound Signal - A system and method for enhancing a tonal sound signal decoded by a decoder of a speech-specific codec in response to a received coded bit stream, in which a spectral analyser is responsive to the decoded tonal sound signal to produce spectral parameters representative of the decoded tonal sound signal. A quantization noise in low-energy spectral regions of the decoded tonal sound signal is reduced in response to the spectral parameters produced by the spectral analyser. The spectral analyser divides a spectrum resulting from spectral analysis into a set of critical frequency bands each comprising a number of frequency bins, and the reducer of quantization noise comprises a noise attenuator that scales the spectrum of the decoded tonal sound signal per critical frequency band, per frequency bin, or per both critical frequency band and frequency bin.02-24-2011
20110288858AUDIO NOISE MODIFICATION FOR EVENT BROADCASTING - An signal processing apparatus, system and software product for audio modification/substitution of a background noise generated during an event including, but not be limited to, substituting or partially substituting a noise signal from one or more microphones by a pre-recorded noise, and/or selecting one or more noise signals from a plurality of microphones for further processing in real-time or near real-time broadcasting.11-24-2011
20130218557Adaptive Approach to Improve G.711 Perceptual Quality - In order to achieve the best improvement of ITU G.711 related codec perceptual quality, perceptual weighting controlling parameter(s) should be at least adaptive to relative quantization error statistics or adaptive to signal level. When the relative quantization error statistics are larger or the signal level is lower, the perceptual weighting should be “stronger”; when the relative quantization error statistics are smaller or the signal level is larger, the perceptual weighting should be “weaker”.08-22-2013
20090164212SYSTEMS, METHODS, AND APPARATUS FOR MULTI-MICROPHONE BASED SPEECH ENHANCEMENT - Systems, methods, and apparatus for processing an M-channel input signal are described that include outputting a signal produced by a selected one among a plurality of spatial separation filters. Applications to separating an acoustic signal from a noisy environment are described, and configurations that may be implemented on a multi-microphone handheld device are also described.06-25-2009
20110144984VOICE CODER WITH TWO MICROPHONE SYSTEM AND STRATEGIC MICROPHONE PLACEMENT TO DETER OBSTRUCTION FOR A DIGITAL COMMUNICATION DEVICE - The present invention provides a voice coder for voice communication that employs a multi-microphone system as part of an improved approach to enhancing signal quality and improving the signal to noise ratio for such voice communications, where there is a special relationship between the position of a first microphone and a second microphone to provide the communication device with certain advantageous physical and acoustic properties. In addition, the communication device can have certain physical characteristics, and design features. In a two microphone arrangement, the first microphone is located in a location directed toward the speech source, while the second microphone is located in a location that provides a voice signal with significantly lower signal-to-noise ratio (SNR).06-16-2011
20120239392SOUND PROCESSING WITH INCREASED NOISE SUPPRESSION - A method for processing sound that includes, generating one or more noise component estimates relating to an electrical representation of the sound and generating an associated confidence measure for the one or more noise component estimates. The method further comprises processing, based on the confidence measure, the sound.09-20-2012
20100100375System and Method for Improved Use of Voice Activity Detection - The present invention is a system and method for packetizing actual noise signals, typically background noise, received by an access gateway from a speaking party and transmitting these packetized noise signals via a network to an egress gateway. The egress gateway converts the packetized noise signal into noise signals suitable for output and transmits the output noise signals to a listening party. When the access gateway detects that no voice signal is being received and only a noise signal is being received for a predetermined period of time, the access gateway instructs the egress network to continually transmit output noise signals to the listening party and ceases to transmit packetized noise signals to the egress gateway.04-22-2010
20100100374APPARATUS AND METHOD FOR VOICE PROCESSING IN MOBILE COMMUNICATION TERMINAL - Disclosed are an apparatus and a method for voice processing in a mobile communication terminal. A plurality of microphones are used to remove environmental noise at the time of voice communication, so that it is possible to perform high-quality voice communication and video telephony. Moreover, it is possible to perform voice recording even when a user does not open a mobile communication terminal. Furthermore, when voice is recorded or sound is recorded during moving image photographing, a plurality of microphones are effectively utilized to achieve good-quality recording and to perform recording conveniently even when the folder or the slider of the mobile communication terminal is closed. Therefore, it is possible to provide improved convenience in using the mobile communication terminal.04-22-2010
20110172997SYSTEMS AND METHODS FOR REDUCING AUDIO NOISE - Various embodiments of systems and methods for reducing audio noise are disclosed. One or more sound components such as noise and network tone can be detected based on power spectrum obtained from a time-domain signal. Results of such detection can be used to make decisions in determination of an adjustment spectrum that can be applied to the power spectrum. The adjusted spectrum can be transformed back into a time-domain signal that substantially removes undesirable noise(s) and/or accounts for known sound components such as the network tone.07-14-2011
20090287482AMBIENT NOISE COMPENSATION SYSTEM ROBUST TO HIGH EXCITATION NOISE - A speech enhancement system controls the gain of an excitation signal to prevent uncontrolled gain adjustments. The system includes a first device that converts sound waves into operational signals. An ambient noise estimator is linked to the first device and an echo canceller. The ambient noise estimator estimates how loud a background noise would be near the first device before or after an echo cancellation. The system then compares the ambient noise estimate to a current ambient noise estimate near the first device to control a gain of an excitation signal.11-19-2009
20090287481SPEECH ENHANCEMENT SYSTEM - A speech enhancement system improves speech conversion within an encoder and decoder. The system includes a first device that converts sound waves into operational signals. A second device selects a template that represents an expected signal model. The selected template models speech characteristics of the operational signals through a speech codebook that is further accessed in a communication channel.11-19-2009
20090271186AUDIO SIGNAL SHAPING FOR PLAYBACK BY AUDIO DEVICES - A technique is provided for limiting distortion of an audio signal being processed for playback by an audio device. In accordance with the technique, the audio signal is compressed to generate a compressed audio signal having a level that does not exceed a compression limit. The compressed audio signal is then soft clipped signal to generate a soft-clipped audio signal having a level that does not exceed a soft clipping limit, wherein the compression limit exceeds the soft clipping limit. The technique may also include passing the audio signal through a shaping filter prior to compressing the audio signal, wherein passing the audio signal through a shaping filter comprises modifying the level of selected frequency components of the audio signal.10-29-2009
20110208518Method of editing a noise-database and computer device - The present invention relates to a method as well as to a computing device (08-25-2011
20120296643GEOTAGGED ENVIRONMENTAL AUDIO FOR ENHANCED SPEECH RECOGNITION ACCURACY - 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.11-22-2012
20080243496Band Division Noise Suppressor and Band Division Noise Suppressing Method - A band division noise suppressor suppressing noise sufficiently with a small amount of processing and a little voice distortion. In the band division noise suppressor, a band dividing section (10-02-2008
20080281589Noise Suppression Device and Noise Suppression Method - There is disclosed a noise suppression device capable of improving the noise suppression accuracy while reducing the audio distortion. In this device, a suppression unit suppresses a noise component from the audio power spectrum by using the detection result of the audio-existing band and the noise band in the audio power spectrum including the noise component. A pitch harmonic structure extracting unit (11-13-2008
20080228474METHODS AND APPARATUS FOR POST-PROCESSING OF SPEECH SIGNALS - Methods and apparatus for post-processing of speech signals are disclosed herein.09-18-2008
20090265168NOISE CANCELLATION SYSTEM AND METHOD - A noise cancellation apparatus includes a noise estimation module for receiving a noise-containing input speech, and estimating a noise therefrom to output the estimated noise; a first Wiener filter module for receiving the input speech, and applying a first Wiener filter thereto to output a first estimation of clean speech; a database for storing data of a Gaussian mixture model for modeling clean speech; and an MMSE estimation module for receiving the first estimation of clean speech and the data of the Gaussian mixture model to output a second estimation of clean speech. The apparatus further includes a final clean speech estimation module for receiving the second estimation of clean speech from the MMSE estimation module and the estimated noise from the noise estimation module, and obtaining a final Wiener filter gain therefrom to output a final estimation of clean speech by applying the final Wiener filter gain.10-22-2009
20080312916Receiver Intelligibility Enhancement System - The intelligibility of speech signals is improved in the many situations where a voice signal is communicated or stored. Means and methods are disclosed for developing a scheme with high voice signal intelligibility without sacrifice of voice quality. The disclosed method comprises certain steps, including, but not limited to: Learning the noise on near-end side and enhancing the far-end voice as a function of the noise level on the near-end side. The disclosed method and apparatus are especially useful to increase the intelligibility of the cell phone's loudspeaker output. The invention includes the processing of an input speech signal to generate an enhanced intelligent signal. In frequency domain, the FFT spectrum of the speech received from the far-end is modified in accordance with the LPC spectrum of the local background noise to generate an enhanced intelligent signal. In time domain, the speech is modified in accordance with the LPC coefficients of the noise to generate an enhanced intelligent signal.12-18-2008
20080270127Speech Recognition Device and Speech Recognition Method - There is provided a voice recognition device and a voice recognition method that enhance the function of noise adaptation processing in voice recognition processing and reduce the capacity of a memory being used. Acoustic models are subjected to clustering processing to calculate the centroid of each cluster and the differential vector between the centroid and each model, model composition between each kind of assumed noise model and the calculated centroid is carried out, and the centroid of each composition model and the differential vector are stored in a memory. In the actual recognition processing, the centroid optimal to the environment estimated by the utterance environmental estimation is extracted from the memory, model restoration is carried out on the extracted centroid by using the differential vector stored in the memory, and noise adaptation processing is executed on the basis of the restored model.10-30-2008
20090125303AUDIO SIGNAL PROCESSING APPARATUS, AUDIO SIGNAL PROCESSING METHOD, AND COMMUNICATION TERMINAL - An audio signal processing apparatus, includes an environmental ambient noise level detection unit for detecting an environmental ambient noise level contained in an audio signal inputted through sound collection means for collecting a transmission sound at the time of a voice call, a signal level adjustment unit which has a level adjustment function to adjust an output signal level with respect to an input signal level, and an input/output characteristic change function to change an input/output characteristic when adjusting a level in the level adjustment function by means of a control signal, and in which a received sound signal in the case of the telephone call voice is arranged to be an input signal, and a control signal generation unit for generating the control signal for changing the input/output characteristic of the signal level adjustment unit from the environmental ambient noise level detected by the environmental ambient noise level detection unit.05-14-2009
20090132244METHOD AND APPARATUS FOR CONTROLLING A VOICE OVER INTERNET PROTOCOL (VoIP) DECODER WITH AN ADAPTIVE JITTER BUFFER - A method and apparatus that controls a Voice over Internet Protocol (VoIP) decoder in a communication device is disclosed. The method may include determining if a packet has been received, and if a packet has been received, determining if a receive error has occurred, and if a receive error has not occurred, setting a counter that counts a number of sequential bad frames to a value of zero, decoding the received packet, and sending the decoded packet to an audio queue for presentation to a user of the communication device.05-21-2009
20090119099System and method for automobile noise suppression - A system and a method for automobile noise suppression in an automobile are provided. The system comprises a processor and a noise suppression device. The noise suppression device is configured for receiving a voice signal, which includes a speech signal and a noise signal. The processor is configured for determining an adjusting parameter set according to an automobile speed signal corresponding to a speed of the automobile. The noise suppression device can suppress the noise signal according to the adjusting parameter set, whereby enhancing the voice quality.05-07-2009
20080306734Signal Noise Reduction - Provision to reduce production of musical noise. A noise reduction device includes: means for calculating a rank for each element included in a first region having predetermined sizes in the time axis direction and in the frequency axis direction, depending on a value of the element, in a noise section of an observed signal indicating variation of a frequency spectrum with time; means for calculating a rank for each element included in a second region, depending on a value of the element, the second region having predetermined sizes in the time axis direction and in the frequency axis direction in the observed signal; and means for subtracting, from the values of the respective elements in the second region, values based on the values of the respective elements in the first region whose ranks correspond to ranks of respective elements in the second region.12-11-2008
20080306733IMAGING APPARATUS, VOICE PROCESSING CIRCUIT, NOISE REDUCING CIRCUIT, NOISE REDUCING METHOD, AND PROGRAM - A noise reducing circuit includes a denoising unit configured to eliminate a noise band from an input voice signal; a noise recognizing unit configured to recognize noise included in the voice signal; a denoising period generating unit configured to generate a signal indicating a denoising period in accordance with an occurrence period of the recognized noise; and a selecting unit configured to select an output of the denoising unit when the denoising period is indicated and select the voice signal when the denoising period is not indicated.12-11-2008
20130218559NOISE REDUCTION APPARATUS, AUDIO INPUT APPARATUS, WIRELESS COMMUNICATION APPARATUS, AND NOISE REDUCTION METHOD - A speech segment of a voice sound is detected based on a first sound pick-up signal obtained based on the voice sound. A voice incoming direction of the voice sound is determined using the first sound pick-up signal and a second sound pick-up signal obtained based on a picked-up sound. A noise reduction process is performed to reduce a noise component carried by the first sound pick-up signal by using the second sound pick-up signal, wherein a noise reduction amount adjusted in accordance with the voice incoming direction is used in the noise reduction process.08-22-2013
20090055170Sound Source Separation Device, Speech Recognition Device, Mobile Telephone, Sound Source Separation Method, and Program - A sound source signal from a target sound source is allowed to be separated from a mixed sound which consists of sound source signals emitted from a plurality of sound sources without being affected by uneven sensitivity of microphone elements. A beamformer section 02-26-2009
20090024387COMMUNICATION SYSTEM NOISE CANCELLATION POWER SIGNAL CALCULATION TECHNIQUES - In order to enhance the quality of a communication signal derived from speech and noise, a filter divides the communication signal into a plurality of frequency band signals. A calculator generates a plurality of power band signals each having a power band value and corresponding to one of the frequency band signals. The power band values are based on estimating, over a time period, the power of one of the frequency band signals. The time period is different for different ones of the frequency band signals. The power band values are used to calculate weighting factors which are used to alter the frequency band signals that are combined to generate an improved communication signal.01-22-2009
20090063142Method and apparatus for controlling echo in the coded domain - A method and corresponding apparatus for coded-domain acoustic echo control is presented. An echo control problem is considered as that of perceptually matching an echo signal to a reference signal. A perceptual similarity function that is based on the coded spectral parameters produced by the speech codec is defined. Since codecs introduce a significant degree of non-linearity into the echo signal, the similarity function is designed to be robust against such effects. The similarity function is incorporated into a coded-domain echo control system that also includes spectrally-matched noise injection for replacing echo frames with comfort noise. Using actual echoes recorded over a commercial mobile network, it is shown herein that the similarity function is robust against both codec non-linearities and additive noise. Experimental results further show that the echo-control is effective at suppressing echoes compared to a Normalized Least Mean Squared (NLMS)-based echo cancellation system.03-05-2009
20090063141Apparatus And Method For Adjusting Prompt Voice Depending On Environment - An apparatus for adjusting a prompt voice depending on an environment comprises a receiver module used for receiving a background sound, an analyzer module generating a control signal according to the background sound and an output module adjusting an output frequency of a prompt voice through the control signal and outputting the adjusted prompt voice.03-05-2009
20110231185METHOD AND APPARATUS FOR BLIND SIGNAL RECOVERY IN NOISY, REVERBERANT ENVIRONMENTS - A maximum-kurtosis, distortionless response (MKDR) technique and an extension, the maximum-kurtosis, Wiener estimate (MKWE) technique, are provided. In one form, blind estimates of the speech source's channel response are made from the microphone data and MVDR is applied. The source direction is estimated by finding weights that maximize output kurtosis, or the fourth central statistical moment, in the frequency domain. The MKWE approach approximates the Wiener filter by using MKDR-output noise power estimates to compute a Wiener post-filter. These approaches can be extended to block-adaptive versions if the speech source is not quickly moving in space.09-22-2011
20120078620Robust Noise Estimation - An enhancement system improves the estimate of noise from a received signal. The system includes a spectrum monitor that divides a portion of the signal at more than one frequency resolution. Adaptation logic derives a noise adaptation factor of the received signal. A plurality of devices tracks the characteristics of an estimated noise in the received signal and modifies multiple noise adaptation rates. Weighting logic applies the modified noise adaptation rates derived from the signal divided at a first frequency resolution to the signal divided at a second frequency resolution.03-29-2012
20090144055Audio Coding System Using Temporal Shape of a Decoded Signal to Adapt Synthesized Spectral Components - A receiver in an audio coding system receives a signal conveying frequency subband signals representing an audio signal. The subband signals are examined to assess one or more characteristics of the audio signal including temporal shape. Spectral components are synthesized having the one or more assessed characteristics, integrated with the subband signals and passed through a synthesis filterbank to generate an output signal.06-04-2009
20090248407SOUND ENCODER, SOUND DECODER, AND THEIR METHODS - A sound encoder enabling prevention of deterioration of the sound quality of a reproduced signal even if the harmonic structure is broken in a part of the sound signal. The filter state position determining section (10-01-2009
20090248409COMMUNICATION APPARATUS - A communication apparatus for adjusting a received voice signal in accordance with an ambient noise, the communication apparatus includes: a microphone for receiving an ambient noise and input voice and outputting a voice input signal corresponding to a level of the input voice and the ambient noise; a receiver for receiving the voice signal; a processer for extracting a voice component originated by a sender and an ambient noise component originated by the ambient noise, determining the ratio between the voice component and the ambient noise component, and adjusting the amplitude of the received voice signal in accordance with the ratio; and a speaker for outputting a reception voice corresponding to the adjusted reception voice signal.10-01-2009
20090248408VOICE EMITTING AND COLLECTING DEVICE - A voice emitting and collecting device that is capable of picking up/outputting a voice emitted from a talker at a high S/N ratio by eliminating the influence of a diffracting voice despite a simple configuration is provided. A signal differencing circuit 10-01-2009
20100179809APPARATUS AND METHOD OF PROCESSING A RECEIVED VOICE SIGNAL IN A MOBILE TERMINAL - An apparatus and a method thereof, processes a voice signal of a mobile terminal in a mobile communication system. The apparatus to process a received-voice signal received through a wireless channel in a mobile terminal includes a digital signal processing unit to generate an encoded packet and frame type information defining a characteristic of the encoded packet by performing voice encoding on an audible signal input from a microphone. The apparatus also includes a received-voice controlling unit to determine a noise level in consideration of the frame type information and a level of the audible signal, and to control at least one of a tone and a volume of received voice by the determined noise level.07-15-2010
20100161324NOISE DETECTION APPARATUS, NOISE REMOVAL APPARATUS, AND NOISE DETECTION METHOD - A noise detection apparatus includes a time-frequency transform unit configured to transform an input signal from a time domain to a frequency domain to produce a spectrum, a power spectrum calculating unit configured to obtain powers of frequencies from the spectrum, a peak stationarity detecting unit configured to use peaks of the powers of frequencies in each frame to detect frequencies at which a stationary peak of the powers exists, a power stationarity detecting unit configured to use magnitudes of the powers of frequencies in each frame to detect frequencies at which the magnitudes of the powers are stationary, and a check unit configured to use the frequencies detected by the peak stationarity detecting unit and the frequencies detected by the power stationarity detecting unit to check whether there is a noise that has at least one of peak stationarity and power stationarity in the frequency domain.06-24-2010
20090259462COMFORT NOISE INFORMATION HANDLING FOR AUDIO TRANSCODING APPLICATIONS - A device comprising an audio information processor to receive at least one audio stream encoded according to a first protocol by a remote network processing device, the audio stream having associated comfort noise information to indicate a level of background noise available for presentation during silence periods associated with the audio stream, the audio information processor to decode the received audio stream according to the first protocol and to encode the decoded audio stream according to a second protocol, and a background noise translator to convert the comfort noise information received with the audio stream into a format compatible with the second protocol.10-15-2009
20120197637SPEECH PROCESSING RESPONSIVE TO A DETERMINED ACTIVE COMMUNICATION ZONE IN A VEHICLE - A system for and method of speech processing for a vehicle. Speech is received from at least one vehicle occupant via a plurality of microphones corresponding to the plurality of zones in the vehicle, wherein the microphones convert the speech into speech signals. At least one active communication zone is determined in which the at least one vehicle occupant corresponding to the active communication zone is speaking Speech processing is modified in response to the determined active communication zone.08-02-2012
20120197636SYSTEM AND METHOD FOR SINGLE-CHANNEL SPEECH NOISE REDUCTION - A system and method may receive a single-channel speech input captured via a microphone. For each current frame of speech input, the system and method may (a) perform a time-frequency transformation on the input signal over L (L>1) frames including the current frame to obtain an extended observation vector of the current frame, data elements in the extended observation vector representing the coefficients of the time-frequency transformation of the L frames of the speech input, (b) compute second-order statistics of the extended observation vector and of noise, and (c) construct a noise reduction filter for the current frame of the speech input based on the second-order statistics of the extended observation vector and the second-order statistics of noise.08-02-2012
20120197638Method and Device for Noise Reduction Control Using Microphone Array - The present invention provides a noise reduction control method using a microphone array and a noise reduction control device using a microphone array wherein the method comprises the steps of: S1: collecting, by the microphone array, acoustic signals; S2: estimating incidence angles of all acoustic signals of the microphone array; S3: conducting a statistics on signal components according to incidence angles; S4: determining a parameter α from a ratio of noise components according to the statistical result and using the parameter α as a control parameter for controlling an adaptive filter. With the present invention, space position information of the sound is obtained directly with the microphone array to control update of the adaptive filter more accurately, so as to eliminate noise, enhance SNR and protect speech quality well at the same time.08-02-2012
20100217586SIGNAL PROCESSING SYSTEM, APPARATUS AND METHOD USED IN THE SYSTEM, AND PROGRAM THEREOF - Provided is a signal separation system including a rendering unit which receives a first and a second input signal and positions the first input signal according to rendering information.08-26-2010
20100217587SIGNAL PROCESSING METHOD AND DEVICE - A signal processor includes: a first adaptive filter that takes a first signal as input and generates a first pseudo signal; a first subtractor that subtracts the first pseudo signal from a second signal to supply a first differential signal as output; a second adaptive filter that takes the first signal as input to generate a second pseudo signal; a second subtractor that subtracts the second pseudo signal from the second signal to supply a second differential signal as output; a first step size control circuit that generates a first step size used in updating the first adaptive filter in accordance with the relation between the second pseudo signal and the second differential signal; and a second step size control circuit that generates a second step size used in updating the second adaptive filter in accordance with the relation between the first signal and the second signal.08-26-2010
20100228545VOICE MIXING DEVICE, NOISE SUPPRESSION METHOD AND PROGRAM THEREFOR - A voice mixing device for mixing a plurality of voice signals, comprises: a speaker selection unit selecting at least one voice signal among said plurality of voice signals; a full signal adder unit adding all of at least one voice signal selected by said speaker selection unit; respective subtractor unit subtracting only one of said selected voice signals from an addition result of said full signal adder unit; a common noise suppression unit suppressing noise of a common voice signal, being an addition result of said full signal adder unit; individual noise suppression unit suppressing noise of respective individual voice signals, being subtraction results of said subtractor unit; and memory switching unit copying information of noise suppression obtained in said common noise suppression unit based on a selection result of said speaker selection unit, to information of noise suppression in said individual noise suppression unit.09-09-2010
20100211387SPEECH PROCESSING WITH SOURCE LOCATION ESTIMATION USING SIGNALS FROM TWO OR MORE MICROPHONES - Computer implemented speech processing is disclosed. First and second voice segments are extracted from first and second microphone signals originating from first and second microphones. The first and second voice segments correspond to a voice sound originating from a common source. An estimated source location is generated based on a relative energy of the first and second voice segments and/or a correlation of the first and second voice segments. A determination whether the voice segment is desired or undesired may be made based on the estimated source location.08-19-2010
20130218558METHOD AND SYSTEM FOR DETECTION OF ONSET OF NEAR-END SIGNAL IN AN ECHO CANCELLATION SYSTEM - A method, a system and a computer program product for fast detection of the onset of a near-end signal is provided. An Acoustic Echo Canceller (AEC) attenuates an acoustic echo present in a tele-communication network. The AEC includes an adaptive filter that estimates the acoustic echo and generates an error signal. The error signal is the difference between the acoustic echo and the estimate of acoustic echo plus a near-end signal, if present. The method comprises computing an onset indicator parameter from the error signal and the estimate of acoustic echo. Several other parameters are subsequently calculated by using the onset indicator parameter, the error signal and the estimate of acoustic echo.08-22-2013
20120143604Method for Restoring Spectral Components in Denoised Speech Signals - Spectral components attenuated in a test denoised speech signal as a result of denoising a test speech signal are restored by representing a training undistorted speech signal as a composition of training undistorted bases, and representing a training denoised speech signal as a composition of training distorted bases. The test denoised signal decomposed as a composition of the training distorted bases. The undistorted test speech signal is then estimated as the composition of the training undistorted bases that is identical to the composition of training distorted bases.06-07-2012
20090254340Noise Reduction - A signal processor for estimating noise power in an audio signal includes a filter unit for generating a series of power values, each power value representing the power in the audio signal at a respective one of a plurality of frequency bands; a signal classification unit for analysing successive portions of the audio signal to assess whether each portion contains features characteristic of speech, and for classifying each portion in dependence on that analysis; a correction unit for estimating a minimum power value in a time-limited part of the audio signal, estimating the total noise power in that part of the audio signal and forming a correction factor dependent on the ratio of the minimum power value to the estimated total noise power, the correction unit being configured to estimate the minimum power value and the total noise power over only those portions of the time-limited part of the signal that are classified by the signal classification unit as being less characteristic of speech; and a noise estimation unit for estimating noise in the audio signal in dependence on the power values output by the filter unit and the correction factor formed by the correction unit.10-08-2009
20090281802SPEECH INTELLIGIBILITY ENHANCEMENT SYSTEM AND METHOD - A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.11-12-2009
20090276212ROBUST DECODER - Techniques and tools related to delayed or lost coded audio information are described. For example, a concealment technique for one or more missing frames is selected based on one or more factors that include a classification of each of one or more available frames near the one or more missing frames. As another example, information from a concealment signal is used to produce substitute information that is relied on in decoding a subsequent frame. As yet another example, a data structure having nodes corresponding to received packet delays is used to determine a desired decoder packet delay value.11-05-2009
20110112831Noise suppression - A method and computing system for suppressing noise in an audio signal, comprising: receiving the audio signal at signal processing means; determining that another signal is input to the signal processing means, the input signal resulting from an activity which generates noise in the audio signal; and selectively suppressing noise in the audio signal in dependence on the determination that the input signal is input to the signal processing means to thereby suppress the generated noise in the audio signal.05-12-2011
20090106021ROBUST TWO MICROPHONE NOISE SUPPRESSION SYSTEM - A system, method, and apparatus for separating speech signal from a noisy acoustic environment. The separation process may include directional filtering, blind source separation, and dual input spectral subtraction noise suppressor. The input channels may include two omnidirectional microphones whose output is processed using phase delay filtering to form speech and noise beamforms. Further, the beamforms may be frequency corrected. The omnidirectional microphones generate one channel that is substantially only noise, and another channel that is a combination of noise and speech. A blind source separation algorithm augments the directional separation through statistical techniques. The noise signal and speech signal are then used to set process characteristics at a dual input noise spectral subtraction suppressor (DINS) to efficiently reduce or eliminate the noise component. In this way, the noise is effectively removed from the combination signal to generate a good qualify speech signal.04-23-2009
20130144617BACKGROUND NOISE CANCELLING DEVICE AND METHOD - A background noise cancelling device for removing a background noise from an input signal in which the background noise is mixed in a voice signal to produce an output signal includes: storage a unit for preliminarily storing a predictable background noise, which is the background noise, as a stored background noise in a state in which a synchronization signal is superimposed on the predictable background noise; an estimation unit for reading the stored background noise from the storage unit and for correlating the read stored background noise and the input signal to establish synchronization by using the synchronization signal and to produce a predicted noise; and a subtracting unit for removing the predicted noise from the input signal to produce the voice signal obtained as a result of the removal.06-06-2013
20110029305METHOD FOR PROCESSING NOISY SPEECH SIGNAL, APPARATUS FOR SAME AND COMPUTER-READABLE RECORDING MEDIUM - A noise estimation method for a noisy speech signal according to an embodiment of the present invention includes the steps of approximating a transformation spectrum by transforming an input noisy speech signal to a frequency domain, calculating a smoothed magnitude spectrum having a decreased difference in a magnitude of the transformation spectrum between neighboring frames, calculating a search spectrum to represent an estimated noise component of the smoothed magnitude spectrum, and estimating a noise spectrum by using a recursive average method using an adaptive forgetting factor defined by using the search spectrum. According to an embodiment of the present invention, the amount of calculation for noise estimation is small, and large-capacity memory is not required. Accordingly, the present invention can be easily implemented in hardware or software. Further, the accuracy of noise estimation can be increase because an adaptive procedure can be performed on each frequency sub-band.02-03-2011
20100250247Method and Apparatus for Speech Signal Processing - A method for speech signal processing is provided. Energy attenuation gain values are set for background noise signals corresponding to obtained background noise frames subsequent to an erasure concealment frame, so that differences between the energy attenuation gain values of the background noise signals corresponding to the background noise frames and the energy attenuation gain values of signals corresponding to their respective previous frames are within a threshold range. Energy attenuation of the background noise signals corresponding to the background noise frames is controlled by using the energy attenuation gain values. An apparatus for speech signal processing is also provided in embodiments of the present invention. By using the embodiments of the present invention, the energy transition between the area of erasure concealment signal and the area of background noise signal may be made natural and smooth, so as to improve the audio comfortable sensation of the listener.09-30-2010
20110246191METHOD, SYSTEM AND PEER APPARATUS FOR IMPLEMENTING MULTI-CHANNEL VOICE MIXING - Embodiments of the present invention provide a method, system and peer apparatus for implementing multi-channel voice mixing, which belongs to a network communication field. The method includes: obtaining, by each peer, voice mixing quality parameters of super peers which are determined from peers according to information processing abilities of the peers; obtaining, by peers with voice input in the peers, priorities of the super peers according to the voice mixing quality parameters, and selecting at least one super peer for voice mixing from all the super peers according to the priorities of the super peers; mixing, by the at least one super peer for voice mixing, audio data of each peer with voice input; and publishing mixed data. The present invention selects a super peer to replace the existing server for implementing multi-channel voice mixing and publishing mixed data. Thus, server costs and bandwidth resources can be saved.10-06-2011
20110246190SPEECH DIALOG APPARATUS - According to one embodiment, a speech dialog apparatus includes a speech detection unit that detects a start and an end of echo removed speech obtained by removing an echo of response speech contained in input speech; a response interruption control unit that outputs a response interruption command if the end is not yet detected when a predetermined period from the detection of the start passes; and a dialog control unit that causes a response speech output unit to interrupt output of the response speech upon receipt of the response interruption command from the response interruption control unit.10-06-2011
20090326934AUDIO DECODING DEVICE, AUDIO DECODING METHOD, PROGRAM, AND INTEGRATED CIRCUIT - An audio decoding device of the present invention includes: a decoding unit decoding a stream to a spectrum coefficient, and outputting stream information when a frame included in the stream cannot be decoded; an orthogonal transformation unit transforming the spectrum coefficient to a time signal; a correction unit generating a correction time signal based on an output waveform within a reference section that is in a section that overlaps between an error frame section to which the stream information is outputted and an adjacent frame section and that is a section in the middle of the adjacent frame section, when the decoding unit outputs the stream information: and an output unit generating the output waveform by synthesizing the correction time signal and the time signal.12-31-2009
20100241426Method and system for noise reduction - Techniques pertaining to noise reduction are disclosed. According to one aspect of the present invention, noise in an audio signal is effectively reduced and a high quality of a target voice is recovered at the same time. In one embodiment, an array of microphones is used to sample the audio signal embedded with noise. The samples are processed according to a beamforming technique to get a signal with an enhanced target voice. A target voice is located in the audio signal sampled by the microphone array. A credibility of the target voice is determined when the target voice is located. The voice presence probability is weighted by the credibility. The signal with the enhanced target voice is enhanced according to the weighed voice presence probability.09-23-2010
20100191527ECHO SUPPRESSING SYSTEM, ECHO SUPPRESSING METHOD, RECORDING MEDIUM, ECHO SUPPRESSOR, SOUND OUTPUT DEVICE, AUDIO SYSTEM, NAVIGATION SYSTEM AND MOBILE OBJECT - An echo suppressing system includes: a sound output device for outputting sound based on a sound signal, including a passing section for allowing passage of a component of a different frequency band, and a plurality of sound output sections, each of which outputs sound based on each of the plurality of sound signals passed through the passing section; a summer for summing the plurality of sound signals to generate a reference sound signal; a sound input device for converting input sound into a sound signal; and an echo suppressor for suppressing echo based on the sound output by the sound output device, including an input section to which a sound signal is input from the sound input device as an observation sound signal, and a correction section for correcting the observation sound signal so as to suppress echo included in the observation sound signal.07-29-2010
20130013304Method and Apparatus for Environmental Noise Compensation - A method of environmental noise compensation a speech audio signal is provided that includes estimating a fast audio energy level and a slow audio energy level in an audio environment, wherein the speech audio signal is not part of the audio environment, and applying a gain to the speech audio signal to generate an environment compensated speech audio signal, wherein the gain is updated based on the estimated slow audio energy level when the estimated fast audio energy level is not indicative of an audio event in the audio environment and the estimated gain is not updated when the estimated fast audio energy level is indicative an audio event in the audio environment.01-10-2013
20100063807SUBTRACTION OF A SHAPED COMPONENT OF A NOISE REDUCTION SPECTRUM FROM A COMBINED SIGNAL - A system and methods of subtraction of a shaped component of a noise reduction spectrum from a combined signal are disclosed. In an embodiment, a method includes identifying a selected frequency component using a corresponding frequency component of a noise sample spectrum. A noise set is comprised of the noise sample spectrum. The method further includes forming a shaped component of a noise reduction spectrum using a processor and a memory based on a combined signal spectrum and the selected frequency component. The method also includes subtracting the shaped component of the noise reduction spectrum from the combined signal spectrum.03-11-2010
20090216526SYSTEM ENHANCEMENT OF SPEECH SIGNALS - A system enhances speech by detecting a speaker's utterance through a first microphone positioned a first distance from a source of interference. A second microphone may detect the speaker's utterance at a different position. A monitoring device may estimate the power level of a first microphone signal. A synthesizer may synthesize part of the first microphone signal by processing the second microphone signal. The synthesis may occur when power level is below a predetermined level.08-27-2009
20110178799METHODS AND SYSTEMS FOR IDENTIFYING SPEECH SOUNDS USING MULTI-DIMENSIONAL ANALYSIS - Methods and systems of identifying speech sound features within a speech sound are provided. The sound features may be identified using a multi-dimensional analysis that analyzes the time, frequency, and intensity at which a feature occurs within a speech sound, and the contribution of the feature to the sound. Information about sound features may be used to enhance spoken speech sounds to improve recognizability of the speech sounds by a listener.07-21-2011
20110178798ADAPTIVE AMBIENT SOUND SUPPRESSION AND SPEECH TRACKING - A device for suppressing ambient sounds from speech received by a microphone array is provided. One embodiment of the device comprises a microphone array, a processor, an analog-to-digital converter, and memory comprising instructions stored therein that are executable by the processor. The instructions stored in the memory are configured to receive a plurality of digital sound signals, each digital sound signal based on an analog sound signal originating at the microphone array, receive a multi-channel speaker signal, generate a monophonic approximation signal of the multi-channel speaker signal, apply a linear acoustic echo canceller to suppress a first ambient sound portion of each digital sound signal, generate a combined directionally-adaptive sound signal from a combination of each digital sound signal by a combination of time-invariant and adaptive beamforming techniques, and apply one or more nonlinear noise suppression techniques to suppress a second ambient sound portion of the combined directionally-adaptive sound signal.07-21-2011
20110077939MODEL-BASED DISTORTION COMPENSATING NOISE REDUCTION APPARATUS AND METHOD FOR SPEECH RECOGNITION - A model-based distortion compensating noise reduction apparatus for speech recognition, includes: a speech absence probability calculator for calculating the probability distribution for absence and existence of a speech using the sound absence and existence information for the frames; a noise estimation updater for estimating a more accurate noise component by updating the variance of the clean speech and noise for each frame; and a speech absence probability-based noise filter for outputting a first clean speech through the speech absence probability transmitted from the speech absence probability calculator and a first noise filter. Further, the model-based distortion compensating noise reduction apparatus includes a post probability calculator for calculating post probabilities for mixtures using a GMM containing a clean speech in the first clean speech; and a final filter designer for forming a second noise filter and outputting an improved final clean speech signal using the second noise filter.03-31-2011
20120303364Encoded packet selection from a first voice stream to create a second voice stream - 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.11-29-2012
20130191118NOISE SUPPRESSING DEVICE, NOISE SUPPRESSING METHOD, AND PROGRAM - Provided is a noise suppressing device including a framing unit that frames an input signal, a band division unit that obtains a band division signal, a band power computation unit that obtains a band power from each band division signal, a noise determination unit that determines whether each band is stationary noise or non-stationary noise, a noise band power estimation unit that estimates a band power of noise of each band, a noise suppression gain decision unit that decides a noise suppression gain of each band, a noise suppression unit that obtains a band division signal whose noise is suppressed, a band synthesis unit that obtains a framed signal whose noise is suppressed, and a frame synthesis unit that obtains an output signal whose noise is suppressed.07-25-2013
20130191117VOICE ACTIVITY DETECTION IN PRESENCE OF BACKGROUND NOISE - 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.07-25-2013
20110153321SYSTEMS AND METHODS FOR IDENTIFYING SPEECH SOUND FEATURES - Systems and methods for detecting features in spoken speech and processing speech sounds based on the features are provided. One or more features may be identified in a speech sound. The speech sound may be modified to enhance or reduce the degree to which the feature affects the sound ultimately heard by a listener. Systems and methods according to embodiments of the invention may allow for automatic speech recognition devices that enhance detection and recognition of spoken sounds, such as by a user of a hearing aid or other device.06-23-2011
20110153320DEVICE AND METHOD FOR ACTIVE NOISE CANCELLING AND VOICE COMMUNICATION DEVICE INCLUDING THE SAME - Provided is a device for active noise cancelling. The active noise cancelling device may include a phase inverter to generate and output a phase-inverted signal from an input voice signal, and an output unit to output the phase-inverted signal to an outside, and to thereby offset the voice signal.06-23-2011
20120123771Method and Apparatus For Wind Noise Detection and Suppression Using Multiple Microphones - 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.05-17-2012
20100004927Speech sound enhancement device - A disclosed speech sound enhancement device includes an SNR calculation unit for calculating an SNR which is a ratio of received speech sound to ambient noise; a first-frequency-range enhancement magnitude calculation unit for calculating, based on the SNR and frequency-range division information indicating a first and a second frequency range, enhancement magnitude of the first frequency range, the first frequency range contributing to an improvement of subjective intelligibility of the received speech sound, the second frequency range contributing to an improvement of subjective articulation of the received speech sound; a second-frequency-range enhancement magnitude calculation unit for calculating enhancement magnitude of the second frequency range based on the enhancement magnitude of the first frequency range; and a spectrum processing unit for processing spectra of the received speech sound using the enhancement magnitude of the first frequency range, the enhancement magnitude of the second frequency range and the frequency-range division information.01-07-2010
20090240495METHODS AND APPARATUS FOR SUPPRESSING AMBIENT NOISE USING MULTIPLE AUDIO SIGNALS - A method for suppressing ambient noise using multiple audio signals may include providing at least two audio signals captured by at least two electro-acoustic transducers. The at least two audio signals may include desired audio and ambient noise. The method may also include performing beamforming on the at least two audio signals in order to obtain a desired audio reference signal that is separate from a noise reference signal.09-24-2009
20110257967Method for Jointly Optimizing Noise Reduction and Voice Quality in a Mono or Multi-Microphone System - The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible.10-20-2011
20080255834Method and Device for Evaluating the Efficiency of a Noise Reducing Function for Audio Signals - A method of evaluating the efficiency of a noise-reducing function adapted to be applied to audio signals and comprising a preliminary step of obtaining a predefined test audio signal X[m] containing a noise-free wanted signal, a noisy signal Xb[m] obtained by adding a predefined noise signal to the test signal X[m], and a processed signal Y[m], obtained by applying the noise-reducing function to the noisy signal Xb[m], is remarkable in that it includes a loudness measuring step (E10-16-2008
20110054889Enhancing Receiver Intelligibility in Voice Communication Devices - The intelligibility of speech signals is improved in the many situations where a voice signal is communicated or stored. Means and methods are disclosed for developing a scheme with high voice signal intelligibility without sacrifice of voice quality. The disclosed method comprises certain steps, including, but not limited to: Learning the noise on near-end side and enhancing the far-end voice as a function of the noise level on the near-end side. The disclosed method and apparatus are especially useful to increase the intelligibility of the cell phone's loudspeaker output. The invention includes the processing of an input speech signal to generate an enhanced intelligent signal. In frequency domain, the FFT spectrum of the speech received from the far-end is modified in accordance with the LPC spectrum of the local background noise to generate an enhanced intelligent signal. In time domain, the speech is modified in accordance with the LPC coefficients of the noise to generate an enhanced intelligent signal.03-03-2011
20110054888DEVICE, METHOD AND SYSTEM FOR DETECTING UNWANTED CONVERSATIONAL MEDIA SESSION - Some embodiments of the invention relate to a method and a system for detecting unwanted conversational media session data. In accordance with one aspect of the invention, a method of detecting unwanted conversation media session data according to some embodiments of the invention may include calculating two or more progressive similarity scores each with respect to a different instant during a progress of a real-time conversational media session, wherein each of said scores is associated with a similarity between the conversational media session's media data that was available at the associated instant and a reference data item corresponding to media data of a previous conversational media session, and evaluating progressive similarity between the real-time conversational media session and the reference data item based upon the two or more progressive similarity scores.03-03-2011
20110022383METHOD FOR PROCESSING NOISY SPEECH SIGNAL, APPARATUS FOR SAME AND COMPUTER-READABLE RECORDING MEDIUM - A sound quality improvement method for a noisy speech signal according to an embodiment of the present invention comprises the steps of estimating a noise signal of an input noisy speech signal by performing a predetermined noise estimation procedure for the noisy speech signal; measuring a relative magnitude difference to represent a relative difference between the noisy speech signal and the estimated noise signal; calculating a modified overweighting gain function with a non-linear structure in which a relatively high gain is allocated to a low-frequency band than a high-frequency band by using the relative magnitude difference; and obtaining an enhanced speech signal by multiplying the noisy speech signal and a time-varying gain function obtained by using the overweighting gain function. Accordingly, the amount of calculation for noise estimation is small, and large-capacity memory is not required. Furthermore, the present invention can be easily implemented in hardware or software, and the accuracy of noise estimation can be increase because an adaptive procedure can be performed on each frequency sub-band.01-27-2011
20110264449Detector and Method for Voice Activity Detection - The embodiments of the present invention relates to a voice activity detector and a method thereof. The voice activity detector is configured to detect voice activity in a received input signal comprising an input section configured to receive a signal from a primary voice detector of said VAD indicative of a primary VAD decision and at least one signal from at least one external VAD indicative of a voice activity decision from the at least one external VAD, a processor configured to combine the voice activity decisions indicated in the received signals to generate a modified primary VAD decision, and an output section configured to send the modified primary VAD decision to a hangover addition unit of said VAD.10-27-2011
20110125494Speech Intelligibility - The perceived quality of a speech signal output from a user apparatus is improved by storing ambient noise profiles each indicating a model power distribution of a respective ambient noise type as a function of frequency; the ambient noise profile at the user apparatus is measured, the measured ambient noise profile is correlated with each of the stored ambient noise profiles, the stored ambient noise profile is selected with which the measured ambient noise profile is most highly correlated, and the speech signal is manipulated in dependence on which of the stored ambient noise profiles is selected, so as to form an improved speech signal.05-26-2011
20110137646Noise Suppression Method and Apparatus - The present invention relates to a method and a filter design apparatus for designing a digital filter arrangement for noise suppression of a signal representing an acoustic recording. The method comprises determining a desired frequency response of the digital filter arrangement. The method is characterised by including a combination of a high pass filter and a noise suppression filter in the filter arrangement. The combination of the high pass filter and the noise suppression filter is selected based on the determined desired frequency response.06-09-2011
20110137647METHOD AND APPARATUS FOR ENHANCING VOICE SIGNAL IN NOISY ENVIRONMENT - A method and apparatus which enhance a voice signal received by a reception terminal from a transmission terminal, the method including: detecting a magnitude of a noise signal peripheral to the reception terminal; checking a volume level which is set in the reception terminal while the voice signal is received from the transmission terminal; and adaptively enhancing at least one of a volume and an articulation of the voice signal on the basis of a magnitude of the noise signal and the checked volume level.06-09-2011
20120310639Wind Noise Reduction - 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.12-06-2012
20120310638AUDIO SIGNAL PROCESSING METHOD, AUDIO APPARATUS THEREFOR, AND ELECTRONIC APPARATUS THEREFOR - 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 signals12-06-2012
20120310636REPLAY APPARATUS, SIGNAL PROCESSING APPARATUS, AND SIGNAL PROCESSING METHOD - A method of selectively performing signal processing in a first mode and in a second mode. In the first mode, a noise cancel signal having a signal characteristic to cancel an external noise component is generated based on a voice signal supplied from a microphone, and an input digital audio signal and the noise cancel signal are combined into a voice signal to be output through a speaker. In the second mode, a sound process for vocal voice is performed on a voice signal supplied from a microphone, a vocal voice component is canceled from a digital audio signal of input music to generate a karaoke signal, and the karaoke signal and the vocal signal are combined into a voice signal to be output through a speaker. The first mode corresponds to an audio replay operation accompanied by noise cancel, and the second mode corresponds to a karaoke operation.12-06-2012
20100063808Spectral Envelope Coding of Energy Attack Signal - MDCT or FFT-based audio coding algorithms often have the problem named here spectral pre-echoes when coding an energy attack signal. This invention presents several possibilities to avoid the spectral pre-echoes existing in decoded signal segment before the energy attack point. The spectral envelope before the attack point can be improved by performing spectrum smoothing, replacing the segment of having spectral pre-echoes or filtering the segment with a combined filter obtained by doing LPC analysis.03-11-2010
20100030556NOISE DETECTING DEVICE AND NOISE DETECTING METHOD - A difference signal calculating unit of a noise detecting device calculates a difference between the amplitudes of a residual signal at each sample timing and a residual signal at the preceding sample timing. A difference signal comparing unit determines whether or not an impulsive noise is present on the basis of the difference signal at the current sample timing, and the difference signal at each sample timing within a predetermined duration from the current sample timing.02-04-2010
20120209602METHOD AND DEVICE FOR PROVIDING A PLURALITY OF AUDIO FILES WITH CONSISTENT LOUDNESS LEVELS BUT DIFFERENT AUDIO CHARACTERISTICS - The invention provides a method and device for enhancing the listening qualities of an audio file by providing the listener with a plurality of modified equalized audio files. Each modified equalized audio file having a consistent loudness level but different audio characteristics. Hence, for an input audio file the current invention allows the listener to individually select the best audio characteristics for them to listen to the content of the input audio file according to their particular requirements without them needing to adjust the loudness level in playback. The invention further enables the listener to switch between the multiple equalized audio files during playback. The invention further includes a SN detector and reducer to eliminate the adverse effects of the presence of sudden, strong noise in the input audio file in the process of generating the plurality of modified equalized audio files.08-16-2012
20120209601Dynamic enhancement of audio (DAE) in headset systems - Various embodiments relate to signal processing and, more particularly, to processing of received speech signals to preserve and enhance speech intelligibility. In one embodiment, a communications apparatus includes a receiving path over which received speech signals traverse in an audio stream, and an dynamic audio enhancement device disposed in the receiving path. The dynamic audio enhancement (“DAE”) device is configured to modify an amount of volume and an amount of equalization of the audio stream. The DAE device can include a noise level estimator (“NLE”) configured to generate a signal representing a noise level estimate. The noise level estimator can include a non-stationary noise detector and a stationary noise detector. The noise level estimator can be configured to generate the signal representing a first noise level estimate based on detection of the non-stationary noise or a second noise level estimate based on detection of the stationary noise.08-16-2012
20120116758Systems and Methods for Enhancing Voice Quality in Mobile Device - Provided are methods and systems for enhancing the quality of voice communications. The method and corresponding system may involve classifying an audio signal into speech, and speech and noise and creating speech-noise classification data. The method may further involve sharing the speech-noise classification data with a speech encoder via a shared memory or by a Least Significant Bit (LSB) of a Pulse Code Modulation (PCM) stream. The method and corresponding system may also involve sharing acoustic cues with the speech encoder to improve the speech noise classification and, in certain embodiments, sharing scaling transition factors with the speech encoder to enable the speech encoder gradually change data rate in the transitions between the encoding modes.05-10-2012
20120010882CONSTRAINED AND CONTROLLED DECODING AFTER PACKET LOSS - A technique is described herein for reducing audible artifacts in an audio output signal generated by decoding a received frame in a series of frames representing an encoded audio signal in a predictive coding system. In accordance with the technique, it is determined if the received frame is one of a predefined number of received frames that follow a lost frame in the series of the frames. Responsive to determining that the received frame is one of the predefined number of received frames, at least one parameter or signal associated with the decoding of the received frame is altered from a state associated with normal decoding. The received frame is then decoded in accordance with the at least one parameter or signal to generate a decoded audio signal. The audio output signal is then generated based on the decoded audio signal.01-12-2012
20120010881Monaural Noise Suppression Based on Computational Auditory Scene Analysis - The present technology provides a robust noise suppression system which may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. An acoustic signal may be received and transformed to cochlear domain sub-band signals. Features such as pitch may be identified and tracked within the sub-band signals. Initial speech and noise models may be then be estimated at least in part from a probability analysis based on the tracked pitch sources. Speech and noise models may be resolved from the initial speech and noise models and noise reduction may be performed on the sub-band signals and an acoustic signal may be reconstructed from the noise-reduced sub-band signals.01-12-2012
20120116759Method, Computer, Computer Program and Computer Program Product for Speech Quality Estimation - The invention relates to a method, computer, computer program and computer program product for speech quality estimation. The method comprises the steps of: determining a coding distortion parameter (Q05-10-2012
20120059649HOWLING CANCELLER - A howling canceller which suppresses occurrence of howling even when an open loop gain exceeds “1” in the whole reproduction band. In the howling canceller, an adaptive filter (03-08-2012
20120029912Hands-free Active Noise Canceling Device - An invention for eliminating the noise generated by a user speaking into a microphonic instrument is disclosed herein. In a first embodiment, the invention comprises a soundproofed housing arranged to cover a user's mouth region and a loudspeaker that outputs a processed sound wave having a phase that is opposite of the user's voice thereby canceling out the user's voice that was confined inside the housing. In a second embodiment, the invention comprises a soundproofed housing arranged to enclose a user's mouth area, a microphone to capture the user's speech, and a loudspeaker that outputs a processed sound wave having a phase that is opposite of the user's voice thereby canceling out the user's voice that was confined inside the housing.02-02-2012
20120059648Voice Activity Detector (VAD) -Based Multiple-Microphone Acoustic Noise Suppression - Acoustic noise suppression is provided in multiple-microphone systems using Voice Activity Detectors (VAD). A host system receives acoustic signals via multiple microphones. The system also receives information on the vibration of human tissue associated with human voicing activity via the VAD. In response, the system generates a transfer function representative of the received acoustic signals upon determining that voicing information is absent from the received acoustic signals during at least one specified period of time. The system removes noise from the received acoustic signals using the transfer function, thereby producing a denoised acoustic data stream.03-08-2012
20120059650METHOD AND DEVICE FOR THE OBJECTIVE EVALUATION OF THE VOICE QUALITY OF A SPEECH SIGNAL TAKING INTO ACCOUNT THE CLASSIFICATION OF THE BACKGROUND NOISE CONTAINED IN THE SIGNAL - A method and device are provided for the objective evaluation of voice quality of a speech signal. The device includes: a module for extracting a background noise signal, referred to as a noise signal, from the speech signal; a module for calculating the audio parameters of the noise signal; a module for classifying the background noise contained in the noise signal on the basis of the calculated audio parameters, according to a predefined set of background noise classes; and a module for evaluating the voice quality of the speech signal on the basis of at least the resulting classification relative to the background noise in the speech signal.03-08-2012
20110093262Active voice cancellation mask - A method and apparatus for transmitting a clear voice signal while effectuating speech privacy and unobtrusiveness through adaptive signal processing; includes a voice input microphone, an electrical line for transmitting representations of the received voice signal from the microphone and having three modulators in it, actuators or speakers spherically disposed about the microphone and incorporated into a mask for creating sound canceling the ambient spatial transmission of the voice inputted into the microphone. The method and apparatus includes means to measure performance, re-introduce user's speech into the earpiece, reduced volume storage, and means to compensate for temperature dependencies. Cancellation actuators or speakers can be incorporated into said mask at various points including the interior mask surface, within the mask structure itself as well as on the exterior surface of the mask.04-21-2011
20120072210SIGNAL PROCESSING METHOD, APPARATUS AND PROGRAM - In one embodiment, a signal processing method is disclosed. The method can perform filter processing of convoluting a tap coefficient in a first signal sequence to generate a second signal sequence. The method can subtract the second signal sequence from a third signal sequence to generate a fourth signal sequence. The third signal sequence includes an echo signal of the first signal sequence. The method can correct the tap coefficient in accordance with an amount of correction determined using a function. The function includes at least one of a first region and a second region, and has values limited. The first region is included in a negative value region of the fourth signal sequence. The second region is included in a positive value region of the fourth signal sequence.03-22-2012
20130185064ECHO REMOVING APPARATUS, ECHO REMOVING METHOD, PROGRAM AND RECORDING MEDIUM - To be provided is an echo removing apparatus including a transmission path estimate update processing unit, and an output selection unit. A fixed section of the transmission path estimate is updated based on an error from an echo estimate determined using all of the fixed section, the holding section, and the update section. These sections are updated depending on whether an estimate obtained by adding the fixed section and the holding section is better than an estimate of the fixed section alone in every fixed period. Only when the estimate is better, the holding section is added to the fixed section cumulatively, and the update section is substituted into the holding section. Depending on whether an estimate is better, an error from the eco estimate determined using all these sections or the fixed section alone is selected as an output.07-18-2013
20120123773System and Method for Multi-Channel Noise Suppression - Described herein are multi-channel noise suppression systems and methods that are configured to detect and suppress wind and background noise using at least two spatially separated microphones: at least one primary speech microphone and at least one noise reference microphone. The multi-channel noise suppression systems and methods are configured, in at least one example, to first detect and suppress wind noise in the input speech signal picked up by the primary speech microphone and, potentially, the input speech signal picked up by the noise reference microphone. Following wind noise detection and suppression, the multi-channel noise suppression systems and methods are configured to perform further noise suppression in two stages: a first linear processing stage that includes a blocking matrix and an adaptive noise canceler, followed by a second non-linear processing stage.05-17-2012
20120123770METHOD AND APPARATUS FOR IMPROVING SOUND QUALITY - Disclosed is a method of improving a sound quality, including: receiving a transmission signal of a first user equipment; removing noise in the transmission signal using noise information of the first user equipment side; performing speech reinforcement with respect to the noise removed transmission signal using noise information of a second user equipment side; and transmitting the speech reinforced transmission signal to the second user equipment.05-17-2012
20090132245Denoising Acoustic Signals using Constrained Non-Negative Matrix Factorization - A method and system denoises a mixed signal. A constrained non-negative matrix factorization (NMF) is applied to the mixed signal. The NMF is constrained by a denoising model, in which the denoising model includes training basis matrices of a training acoustic signal and a training noise signal and statistics of weights of the training basis matrices. The applying produces weight of a basis matrix of the acoustic signal, of the mixed signal. A product of the weights of the basis matrix of the acoustic signal and the training basis matrices of the training acoustic signal and the training noise signal is taken to reconstruct the acoustic signal. The mixed signal can be speech and noise.05-21-2009
20120166188SELECTIVE NOISE FILTERING ON VOICE COMMUNICATIONS - Embodiments of the invention may provide the ability to selective filter sound from a voice communication, such as a telephone call, based on one or more attributes of the voice communication. Embodiments of the invention may select a filtering profile corresponding to the one or more attributes, and filter sound from the voice communication according to the selected profile. In one embodiment of the invention, the one or more attributes of the voice communication are determined from an electronic record corresponding to the voice communication, such as a calendar entry.06-28-2012
20100204986SYSTEMS AND METHODS FOR RESPONDING TO NATURAL LANGUAGE SPEECH UTTERANCE - Systems and methods for receiving natural language queries and/or commands and execute the queries and/or commands. The systems and methods overcome the deficiencies of prior art speech query and response systems through the application of a complete speech-based information query, retrieval, presentation and command environment. This environment makes significant use of context, prior information, domain knowledge, and user specific profile data to achieve a natural environment for one or more users making queries or commands in multiple domains. Through this integrated approach, a complete speech-based natural language query and response environment can be created. The systems and methods creates, stores and uses extensive personal profile information for each user, thereby improving the reliability of determining the context and presenting the expected results for a particular question or command.08-12-2010
20100082339Wind Noise Reduction - 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.04-01-2010
20120259626INTEGRATED PSYCHOACOUSTIC BASS ENHANCEMENT (PBE) FOR IMPROVED AUDIO - Psychoacoustic Bass Enhancement (PBE) is integrated with one or more other audio processing techniques, such as active noise cancellation (ANC), and/or receive voice enhancement (RVE), leveraging each technique to achieve improved audio output. This approach can be advantageous for improving the performance of headset speakers, which often lack adequate low-frequency response to effectively support ANC.10-11-2012
20100274561Noise Suppression Method and Apparatus - The present invention relates to a method and apparatus of a digital filter for noise suppression of a signal representing an acoustic recording. The method comprises determining a desired frequency response (H(ω)) of the digital filter; and generating a noise suppression filter based on the desired frequency response. The desired frequency response is determined in a manner so that the desired frequency response does not exceed a maximum level, wherein the maximum level is determined in response to the signal to be filtered.10-28-2010
20100274560SELECTIVE RESOLUTION SPEECH PROCESSING - A hearing prosthesis, including receiver means for receiving a signal representative of a sound signal over a frequency range; a first filter bank, having a relatively higher resolution, adapted to process said received signal and produce a first set of channel outputs relating to a selected region or regions of said frequency range; and a second filter bank having a relatively lower resolution, adapted to process said received signal and produce a second set of channel outputs relating to at least the rest of said frequency range; combination means to combine the first and second sets of channel outputs, and processing means operative upon the combined outputs so as to produce a set of stimulation signals for said hearing prosthesis.10-28-2010
20120179462System and Method for Adaptive Intelligent Noise Suppression - Systems and methods for adaptive intelligent noise suppression are provided. In exemplary embodiments, a primary acoustic signal is received. A speech distortion estimate is then determined based on the primary acoustic signal. The speech distortion estimate is used to derive control signals which adjust an enhancement filter. The enhancement filter is used to generate a plurality of gain masks, which may be applied to the primary acoustic signal to generate a noise suppressed signal.07-12-2012
20120179461METHOD FOR JOINTLY OPTIMIZING NOISE REDUCTION AND VOICE QUALITY IN A MONO OR MULTI-MICROPHONE SYSTEM - The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible.07-12-2012
20120123772System and Method for Multi-Channel Noise Suppression Based on Closed-Form Solutions and Estimation of Time-Varying Complex Statistics - Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system.05-17-2012
20120232890APPARATUS AND METHOD FOR DISCRIMINATING SPEECH, AND COMPUTER READABLE MEDIUM - According to one embodiment, an apparatus for discriminating speech/non-speech of a first acoustic signal includes a weight assignment unit, a feature extraction unit, and a speech/non-speech discrimination unit. The first acoustic signal includes a user's speech and a reproduced sound. The reproduced sound is a system sound having a plurality of channels reproduced from a plurality of speakers. The weight assignment unit is configured to assign a weight to each frequency band based on the system sound. The feature extraction unit is configured to extract a feature from a second acoustic signal based on the weight of each frequency band. The second acoustic signal is the first acoustic signal in which the reproduced sound is suppressed. The speech/non-speech discrimination unit is configured to discriminate speech/non-speech of the first acoustic signal based on the feature.09-13-2012
20120253798Rejecting Noise with Paired Microphones - A system for combining signals includes a first microphone generating a first input signal having a first voice component and a first noise component, a second microphone generating a second input signal having a second voice component and a second noise component, a mixing circuit, and an adaptive filter. The mixing circuit applies a first gain having a value α to the first input signal to produce a first scaled signal, applies a second gain having a value 1−α to the second input signal to produce a second scaled signal, and sums the first scaled signal and the second scaled signal to produce a summed signal. The adaptive filter computes an updated value of α to minimize the energy of the summed signal based on the summed signal, the first input signal and the second input signal, and provides the updated value of α to the mixing circuit.10-04-2012
20120316869GENERATING A MASKING SIGNAL ON AN ELECTRONIC DEVICE - 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.12-13-2012
20110004471METHOD AND MEANS FOR ENCODING BACKGROUND NOISE INFORMATION - The inventive method provides for an encoder in a voice codec to be designed such that after a particular idle time (“Idle Period”) it recalculates the averaged energy and the autocorrelation function. Administrative points in the network inform the encoder about the idle time which has been set in the transmission network.01-06-2011
20110004470Method for Wind Noise Reduction - A noisy signal is picked up by a microphone, digitized by an Analog to Digital Converter and fed to a processor for analysis and wind noise reduction. Most of noise reduction methods are based on the assumption that the interfering noise is stationary or slowly varying compared with speech. This assumption allows “learning” the characteristics of the noise between speech pauses and, based on a noise estimate, to build different filters that reduce the noise. In the case of wind noise this basic assumption is not valid. Wind noise is highly non-stationary, its power and spectral characteristics vary greatly. Because wind noise is not stationary, regular noise reduction methods cannot be used to reduce wind noise. For reducing wind noise effects in a device, the presence of wind should be detected reliably and then a novel approach presented here must be applied to eliminate the wind noise.01-06-2011
20120330653DEVICE AND METHOD FOR CAPTURING AND PROCESSING VOICE - A portable voice capture device comprising: 12-27-2012
20120278070COMBINED MICROPHONE AND EARPHONE AUDIO HEADSET HAVING MEANS FOR DENOISING A NEAR SPEECH SIGNAL, IN PARTICULAR FOR A " HANDS-FREE" TELEPHONY SYSTEM - The headset comprises: a physiological sensor suitable for being coupled to the cheek or the temple of the wearer of the headset and for picking up non-acoustic voice vibration transmitted by internal bone conduction; lowpass filter means for filtering the signal as picked up; a set of microphones picking up acoustic voice vibration transmitted by air from the mouth of the wearer of the headset; highpass filter means and noise-reduction means for acting on the signals picked up by the microphones; and mixer means for combining the filtered signals to output a signal representative of the speech uttered by the wearer of the headset. The signal of the physiological sensor is also used by means for calculating the cutoff frequency of the lowpass and highpass filters and by means for calculating the probability that speech is absent.11-01-2012
20120330652SPACE-TIME NOISE REDUCTION SYSTEM FOR USE IN A VEHICLE AND METHOD OF FORMING SAME - 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.12-27-2012
20120290296Method, Apparatus, and Computer Program for Suppressing Noise - 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.11-15-2012
20130013303Processing Audio Signals - 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.01-10-2013
20130179160System and Method for Improved Use of Voice Activity - The present invention is a system and method for packetizing actual noise signals, typically background noise, received by an access gateway from a speaking party and transmitting these packetized noise signals via a network to an egress gateway. The egress gateway converts the packetized noise signal into noise signals suitable for output and transmits the output noise signals to a listening party. When the access gateway detects that no voice signal is being received and only a noise signal is being received for a predetermined period of time, the access gateway instructs the egress network to continually transmit output noise signals to the listening party and ceases to transmit packetized noise signals to the egress gateway.07-11-2013
20130144616SYSTEM AND METHOD FOR MACHINE-MEDIATED HUMAN-HUMAN CONVERSATION - 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.06-06-2013
20130096914System And Method For Utilizing Inter-Microphone Level Differences For Speech Enhancement - Systems and methods for utilizing inter-microphone level differences to attenuate noise and enhance speech are provided. In exemplary embodiments, energy estimates of acoustic signals received by a primary microphone and a secondary microphone are determined in order to determine an inter-microphone level difference (ILD). This ILD in combination with a noise estimate based only on a primary microphone acoustic signal allow a filter estimate to be derived. In some embodiments, the derived filter estimate may be smoothed. The filter estimate is then applied to the acoustic signal from the primary microphone to generate a speech estimate.04-18-2013
20100280826SOUND SOURCES SEPARATION AND MONITORING USING DIRECTIONAL COHERENT ELECTROMAGNETIC WAVES - An apparatus and a method that achieve physical separation of sound sources by pointing directly a beam of coherent electromagnetic waves (i.e. laser). Analyzing the physical properties of a beam reflected from the vibrations generating sound source enable the reconstruction of the sound signal generated by the sound source, eliminating the noise component added to the original sound signal. In addition, the use of multiple electromagnetic waves beams or a beam that rapidly skips from one sound source to another allows the physical separation of these sound sources. Aiming each beam to a different sound source ensures the independence of the sound signals sources and therefore provides full sources separation.11-04-2010
20080201137Method of estimating noise levels in a communication system - A method of estimating noise in data containing voice information and noise includes receiving the data as a sequence of input values; transforming the data by applying a first non linear mapping to the input values wherein the derivative function of the mapping decreases in magnitude as the input values increase in magnitude smoothing the transformed data; and transforming the smoothed transformed data by applying a second non linear mapping that is opposite to the first non linear mapping, to determine an estimate of the noise in the inputted data.08-21-2008
20080300869Audio Signal Dereverberation - A method of estimating the reverberations in an acoustic signal (y) comprises the steps of determining the frequency spectrum (Y) of the signal (y), providing a first parameter (α) indicative of the decay of the reverberations part (r) of the signal over time, and providing a second parameter (β) indicative of the amplitude of the direct part (d) of the signal relative to the reverberations part (r). An estimated frequency spectrum ({hacek over (R)}) of the reverberations signal (r) is produced using the frequency spectrum (Y) of a previous frame, the first parameter (α), and the second parameter (β). The second parameter (β). The second parameter (β) is preferably inversely proportional to the early-to-late ratio of the signal (y).12-04-2008
20130151247METHOD AND DEVICE FOR SUPPRESSING RESIDUAL ECHOES - The present invention discloses a method and a device for suppressing residual echoes. The method comprises: performing adaptive filtering on M transmitter signals respectively to obtain M adaptive filtered signals; performing array-filtering on the M−1 adaptive filtered signals other than the first adaptive filtered signal to obtain M−1 array-filter output signals; subtracting each of the M−1 array-filter output signals from the first adaptive filtered signal respectively to obtain M−1 difference signals, performing time-domain/frequency-domain conversion on the M−1 difference signals respectively and selecting one of the frequency-domain signals that has the least energy; performing time-domain/frequency-domain conversion on the first adaptive filtered signal and the M06-13-2013
20130138434NOISE SUPPRESSION DEVICE - A noise suppression device includes: a power spectrum calculator converting an input signal of time domain into power spectra of frequency domain; a voice/noise determination unit determining whether the power spectra indicate voice or noise; a noise spectrum estimation unit estimating noise spectra of the power spectra; a period component estimation unit analyzing a harmonic structure constituting the power spectra and estimating periodical information about the power spectra; a weighting coefficient calculator calculating a weighting coefficient for weighting the power spectra; a suppression coefficient calculator calculating a suppression coefficient for suppressing noise included in the power spectra; a spectrum suppression unit suppressing amplitude of the power spectra in accordance with the suppression coefficient; and an inverse Fourier transformer converting the power spectra output by the spectrum suppression unit into a signal of time domain to generate a noise-suppressed signal.05-30-2013
20090210221COMMUNICATION SYSTEM FOR BUILDING SPEECH DATABASE FOR SPEECH SYNTHESIS, RELAY DEVICE THEREFOR, AND RELAY METHOD THEREFOR - A relay device 08-20-2009
20100318352METHOD AND MEANS FOR ENCODING BACKGROUND NOISE INFORMATION - The invention relates to a method and means for encoding background noise information during voice signal encoding methods. A basic idea of the invention is to provide the scalability known for transmitting voice information in a similar manner when forming an SID frame. The invention provides encoding of a narrowband first component and of a broadband second component of a piece of background noise information and formation of an SID frame which describes the background noise with separate areas for the first and second components.12-16-2010
20130158989APPARATUS AND METHOD FOR NOISE REMOVAL - 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.06-20-2013
20130158990INFORMATION PROCESSING APPARATUS AND PROGRAM - According to one embodiment, an information processing apparatus includes a first signal input unit configure to receive a first signal, a second signal input unit configure to receive a signal, a first control unit configure to acquire system resources, a second control unit configure to select, in accordance with information of the system resources acquired by the first control unit, a processing method for suppressing at least one of echo and noise of the second signal input from the second signal input unit containing the echo due to the first signal input from the first signal input unit, a third control unit configure to generate an output signal by suppressing at least one of the echo and the noise from the second signal by the processing method selected by the second control unit, and a signal output unit configure to output the output signal generated by the third control unit.06-20-2013
20110307249METHOD AND ACOUSTIC SIGNAL PROCESSING SYSTEM FOR INTERFERENCE AND NOISE SUPPRESSION IN BINAURAL MICROPHONE CONFIGURATIONS - A method determines a bias reduced noise and interference estimation in a binaural microphone configuration with a right and a left microphone signal at a time-frame with a target speaker active. The method includes a determination of the auto power spectral density estimate of the common noise formed of noise and interference components of the right and left microphone signals and a modification of the auto power spectral density estimate of the common noise by using an estimate of the magnitude squared coherence of the noise and interference components contained in the right and left microphone signals determined at a time frame without a target speaker active. An acoustic signal processing system and a hearing aid implement the method for determining the bias reduced noise and interference estimation. The noise reduction performance of speech enhancement algorithms is improved by the invention. Further, distortions of the target speech signal and residual noise and interference components are reduced.12-15-2011
20130191119SIGNAL PROCESSING DEVICE, SIGNAL PROCESSING METHOD AND SIGNAL PROCESSING PROGRAM - From a mixed signal in which a first signal and a second signal are mixed, the second signal is removed at low processing cost and without delay. As a result, an estimated first signal which has low residue of the second signal and low distortion is obtained.07-25-2013
20120029915METHOD FOR PROCESSING MULTICHANNEL ACOUSTIC SIGNAL, SYSTEM THEREFOR, AND PROGRAM - A method for processing multichannel acoustic signals which processes input signals of a plurality of channels including the voices of a plurality of speaking persons. The method is characterized by detecting the voice section of each speaking person or each channel, detecting overlapped sections wherein the detected voice sections are common between channels, determining a channel to be subjected to crosstalk removal and the section thereof by use of at least voice sections not including the detected overlapped sections, and removing crosstalk in the sections of the channel to be subjected to the crosstalk removal.02-02-2012
20120029914Method and apparatus for transmitting wideband speech signals - A method and an apparatus for transmitting a speech signal are provided. A speech signal transmitter includes a quadrature mirror filter, a base sub-band encoder, an enhancement sub-band encoder, and a network connector. The quadrature mirror filter receives a speech signal, divides the speech signal into an enhancement band speech signal and a base band speech signal, and outputs the enhancement band speech signal and the base band speech signal. The base sub-band encoder receives and encodes the base band speech signal. The enhancement sub-band encoder receives and encodes the enhancement band speech signal. The network connector multiplexes the encoded enhancement band speech signal and the encoded base band speech signal based on the kinds of networks over which speech signals are transmitted, and transmits the multiplexed signals to the networks. A speech signal is multiplexed and transmitted by various methods based on the kinds of networks. Thus, the speech signal can be efficiently transmitted.02-02-2012
20120029913Sound Quality Control Apparatus and Sound Quality Control Method - According to one embodiment, there is provided a sound quality control apparatus, including: a characteristic parameter extractor; a speech score calculator; a music score calculator; a power value acquisition module; a first storage configured to store speech scores and music scores; a second storage configured to store power values; a power-based score corrector configured to correct a current music score or a current speech score based on a first comparison result between a current power value and past power values, a second comparison result between the current music score and past music scores and a third comparison result between the current speech score and past speech scores; and a sound quality controller configured to perform a sound quality control by using at least one of the speech score and the music score corrected by the power-based score corrector.02-02-2012
20130197904Indirect Model-Based Speech Enhancement - 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.08-01-2013
20130197905SIGNAL PROCESSING DEVICE, SIGNAL PROCESSING METHOD AND SIGNAL PROCESSING PROGRAM - To achieve sufficient noise cancellation when a reference signal cannot be captured in the proximity of a noise source.08-01-2013
20120035921Dynamic Noise Reduction - A speech enhancement system improves the speech quality and intelligibility of a speech signal. The system includes a time-to-frequency converter that converts segments of a speech signal into frequency bands. A signal detector measures the signal power of the frequency bands of each speech segment. A background noise estimator measures a background noise detected in the speech signal. A dynamic noise reduction controller dynamically models the background noise in the speech signal. The speech enhancement renders a speech signal perceptually pleasing to a listener by dynamically attenuating a portion of the noise that occurs in a portion of the spectrum of the speech signal.02-09-2012
20120035920NOISE ESTIMATION APPARATUS, NOISE ESTIMATION METHOD, AND NOISE ESTIMATION PROGRAM - A noise estimation apparatus includes a correlation calculator configured to calculate a correlation value of a spectrum between a plurality of frames in sound information obtained using one or more microphones, a power calculator configured to calculate a power value indicating a sound level of one target frame among the plurality of frames, an update determiner configured to determine an update degree indicating a degree to which the sound information of the target frame is to be reflected in a noise model stored in a storage, or determine whether or not the noise model is to be updated to another noise model, based on the power value of the target frame and the correlation value, and an updater configured to generate the other noise model based on a determined result, the sound information of the target frame, and the noise model.02-09-2012
20130204616Computer-Implemented System and Method for Enhancing Audio to Individuals Participating in a Conversation - A computer-implemented system and method for enhancing audio to individuals participating in a conversation is provided. Audio data for individuals participating in one or more conversations is analyzed. Possible conversational configurations of the individuals are generated based on the audio data, and each possible conversational configuration includes one or more subconfigurations of at least two of the individuals. A probability weight is assigned to each of the subconfigurations and includes a likelihood that the individuals of that subconfiguration are participating in one of the conversations. A probability of each possible conversational configuration is determined by combining the probability weights for the subconfigurations of that possible conversational configuration. The possible conversational configuration with the highest probability is selected as a most probable configuration. The individuals participating in the conversations are determined based on the most probable configuration. Audio for each individual participating in the determined conversations is enhanced.08-08-2013
20120084083METHOD AND APPARATUS FOR PROCESSING AUDIO SIGNAL IN A MOBILE COMMUNICATION TERMINAL - A method and an apparatus for processing an audio signal in a mobile terminal are provided, wherein an audio signal received from a counterpart mobile terminal is classified into a voice signal and a noise signal according to respective energy, and a frequency of the classified voice signal and an energy of the classified noise signal is controlled according to a predetermined criteria, then the controlled voice signal and the controlled noise signal are coupled and output to a speaker.04-05-2012

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