Speech Enhancement, E.g., Noise Reduction, Echo Cancellation, Etc. (epo) Patents (Class 704/E21.002)
E Subclasses
- Speech corrupted by noise (EPO) (Class 704/E21.004)
- Speech corrupted by echo-reverberation (EPO) (Class 704/E21.007)
- Speech corrupted by stress-Lombard effect (EPO) (Class 704/E21.008)
- Enhancement of intelligibility of clean or coded speech (EPO) (Class 704/E21.009)
- Separate reconstruction of interference and of speech signal (EPO) (Class 704/E21.012)
- Active noise canceling (EPO) (Class 704/E21.014)
- Public address system (EPO) (Class 704/E21.015)
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Publication number: 20120290296Abstract: 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.Type: ApplicationFiled: June 25, 2012Publication date: November 15, 2012Applicant: NEC CORPORATIONInventors: Akihiko Sugiyama, Masanori Katou
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Publication number: 20120278070Abstract: 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.Type: ApplicationFiled: April 18, 2012Publication date: November 1, 2012Applicant: PARROTInventors: Michael Herve, Guillaume Vitte
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Publication number: 20120263302Abstract: A voice transmission apparatus is provided. The voice transmission apparatus includes a voice input unit, an Analog-to-Digital (AD) converter, a transmission synchronization unit, a channel status determination unit, and a transmission unit. The voice input unit receives an analog voice signal corresponding to a communication session. The AD converter converts the analog voice signal into a digital signal. The transmission synchronization unit generates a transmission target signal by respectively assigning a start identifier and an end identifier to the first frame and final frame of the digital signal corresponding to the communication session. The channel status determination unit determines the status of a channel. The transmission unit transmits the transmission target signal over the channel in such a way that the transmission rate of the transmission target signal is set on a communication session basis depending on the status of the channel.Type: ApplicationFiled: April 6, 2012Publication date: October 18, 2012Applicant: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTEInventors: Young-Ho SON, Jang-Hong YOON
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Publication number: 20120259626Abstract: 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.Type: ApplicationFiled: December 15, 2011Publication date: October 11, 2012Applicant: QUALCOMM IncorporatedInventors: Ren Li, Pei Xiang
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Publication number: 20120259625Abstract: An adaptive audio system can be implemented in a communication device. The adaptive audio system can enhance voice in an audio signal received by the communication device to increase intelligibility of the voice. The audio system can adapt the audio enhancement based at least in part on levels of environmental content, such as noise, that are received by the communication device. For higher levels of environmental content, for example, the audio system might apply the audio enhancement more aggressively. Additionally, the adaptive audio system can detect substantially periodic content in the environmental content. The adaptive audio system can further adapt the audio enhancement responsive to the environmental content.Type: ApplicationFiled: June 18, 2012Publication date: October 11, 2012Applicant: SRS LABS, INC.Inventors: Jun Yang, Richard J. Oliver, James Tracey, Xing He
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Publication number: 20120253798Abstract: 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 ? to the first input signal to produce a first scaled signal, applies a second gain having a value 1?? 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 ? 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 ? to the mixing circuit.Type: ApplicationFiled: April 1, 2011Publication date: October 4, 2012Inventors: Luke C. Walters, Vasu Iyengar, Martin David Ring
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Publication number: 20120250913Abstract: Electronic devices and accessories are provided that may communicate over wired communications paths. The electronic devices may be portable electronic devices such as cellular telephones or media players and may have audio connectors such as 3.5 mm audio jacks. The accessories may be headsets or other equipment having mating 3.5 mm audio plugs and speakers for playing audio. Microphones may be included in an accessory to gather voice signals and noise cancellation signals. Analog-to-digital converter circuitry in the accessory may digitize the microphone signals. Digital voice signals and voice noise cancellation signals can be transmitted over the communications path and processed by audio digital signal processor circuitry in an electronic device. Digital-to-analog converter circuitry in the accessory may convert digital audio signals to analog speaker signals.Type: ApplicationFiled: June 12, 2012Publication date: October 4, 2012Inventors: Wendell B. Sander, Jeffrey J. Terlizzi, Brian Sander, David Tupman, Barry Corlett
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Publication number: 20120239392Abstract: 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.Type: ApplicationFiled: November 1, 2011Publication date: September 20, 2012Inventors: Stefan J. Mauger, Adam A. Hersbach, Pam W. Dawson, John M. Heasman
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Publication number: 20120226495Abstract: A device and a method for filtering out noise from speech of caller are disclosed. The method is applied to the device, includes: inputting a speech sound of a caller; converting the speech sound to digital signals by an analyzing-to-digital converting unit; analyzing the digital signals to identify a pure speech of the caller and filtering out an extraneous noise thus obtaining pure speech signals of the caller; encoding the pure speech signals by a coder and decoder unit, and submitting the encoded speech signals to the receiver.Type: ApplicationFiled: September 23, 2011Publication date: September 6, 2012Applicants: HON HAI PRECISION INDUSTRY CO., LTD., FU TAI HUA INDUSTRY (SHENZHEN) CO., LTD.Inventors: WEI WU, XIN YANG
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Publication number: 20120221329Abstract: A method of speech enhancement in a room (10) includes the steps of capturing audio signals from a speaker's voice by a microphone (12), estimating an ambient noise level in the room from the captured audio signals, processing the captured audio signals by an audio signal processing unit (20), estimating a reverberation level, determining the gain to be applied to the captured audio signals by the audio signal processing unit according to a comparison between the estimated ambient noise level and the estimated reverberation level, and generating sound according to the processed audio signals by a loudspeaker arrangement (24) located in the room, wherein the reverberation level is the level of reverberant components of the sound generated by the loudspeaker arrangement.Type: ApplicationFiled: October 27, 2009Publication date: August 30, 2012Applicant: Phonak AGInventor: Samuel Harsch
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Publication number: 20120221327Abstract: A method, a device and a system for voice encoding/decoding are disclosed in the present invention. The method includes: assembling an input pulse code modulation signal into one signal according to a designated time slot and assembly manner; and encoding the assembled signal according to a designated encoding manner to output an encoded voice signal. In the present invention, because a process of assembling or splitting the signal may be implemented through software, in the case that hardware in a current network does not need to be replaced, an effect of encoding/decoding voice with a 7 K spectrum may be achieved in the current network.Type: ApplicationFiled: May 4, 2012Publication date: August 30, 2012Applicant: Huawei Technologies Co., Ltd.Inventors: Xiaoshuang Li, Xingguo Gao
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Publication number: 20120221328Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between segments of the audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.Type: ApplicationFiled: May 3, 2012Publication date: August 30, 2012Applicant: DOLBY LABORATORIES LICENSING CORPORATIONInventor: Hannes Muesch
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Publication number: 20120215529Abstract: A method for processing and iteratively enhancing and estimating a source audio signal received at two audio receivers is provided. In one embodiment, the method involves the use of codebook constrained iterative binaural Wiener filter (CCIBWF). The provided CCIBWF embodiment can improve the quality of speech received at two audio receivers both in terms of noise reduction and speech intelligibility. In one embodiment, optimum speech enhancement performance was achieved within two iterations of the CCIBWF scheme. Further, the embodiment of the CCIBWF scheme introduces minimal distortion to the binaural cues, such as the interaural time delay cues, thereby preserving localization information of the audio source. The embodiment of the CCIBWF is also able to relatively accurately track the Time Delay of Arrival (TDOA) when the audio source is moving. This ensures that the performance of the CCIBWF scheme is not significantly degraded due to the selection of wrong codebooks.Type: ApplicationFiled: November 2, 2010Publication date: August 23, 2012Applicant: INDIAN INSTITUTE OF SCIENCEInventors: Nadir Cazi, Thippur Venkatanarasaiah Sreenivas
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Publication number: 20120215530Abstract: A method of speech enhancement in a room (10), having the steps of: determining acoustic parameters of the room and a loudspeaker arrangement (24) located in the room, capturing audio signals from a speaker's voice with a microphone (12), and processing the captured audio signals with an audio signal processing unit (20). The audio signals are filtered by applying a selected frequency response curve to the audio signals, generating sound according to the processed audio signals by the loudspeaker arrangement, determining a value indicative of the overall gain applied to the captured audio signals, and selecting a frequency response curve to be applied to the captured audio signals according to the overall gain value and the acoustic parameters.Type: ApplicationFiled: October 27, 2009Publication date: August 23, 2012Applicant: PHONAK AGInventor: Samuel Harsch
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Publication number: 20120209602Abstract: 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.Type: ApplicationFiled: April 16, 2012Publication date: August 16, 2012Applicant: Nuance Communications, Inc.Inventor: Patrick Naylor
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Publication number: 20120209601Abstract: 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.Type: ApplicationFiled: January 9, 2012Publication date: August 16, 2012Inventor: Zhinian Jing
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Publication number: 20120197636Abstract: 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.Type: ApplicationFiled: February 1, 2011Publication date: August 2, 2012Inventors: Jacob Benesty, Yiteng Huang
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Publication number: 20120197637Abstract: 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.Type: ApplicationFiled: January 31, 2012Publication date: August 2, 2012Applicant: GM GLOBAL TECHNOLOGY OPERATIONS, LLCInventors: Jesse T. Gratke, Gary M. Buch, Nathan D. Ampunan, Douglas C. Martin, Bassam S. Shahmurad
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Publication number: 20120197638Abstract: 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 ? from a ratio of noise components according to the statistical result and using the parameter ? 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.Type: ApplicationFiled: December 15, 2010Publication date: August 2, 2012Applicant: GOERTEK INC.Inventors: Bo Li, Shasha Lou, Song Li
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Publication number: 20120197639Abstract: A system improves speech detection or processing by identifying registration signals. The system encodes a limited frequency band by varying the amplitude of a pulse width modulated signal between predefined values. The signal is separated into frequency bins that identify amplitude and phase. The registration signal is measured by comparing a difference in average acoustic power in a plurality of adjacent bins over time.Type: ApplicationFiled: April 11, 2012Publication date: August 2, 2012Inventors: Mark Fallat, Derek Sahota
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Patent number: 8233636Abstract: 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.Type: GrantFiled: August 28, 2006Date of Patent: July 31, 2012Assignee: NEC CorporationInventors: Akihiko Sugiyama, Masanori Katou
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Publication number: 20120191447Abstract: 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.Type: ApplicationFiled: January 24, 2011Publication date: July 26, 2012Applicant: CONTINENTAL AUTOMOTIVE SYSTEMS, INC.Inventors: Bijal Joshi, Suat Yeldener
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Publication number: 20120185246Abstract: 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.Type: ApplicationFiled: July 1, 2011Publication date: July 19, 2012Applicant: Broadcom CorporationInventors: Xianxian Zhang, Jes Thyssen, Kwan Young Shin
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Publication number: 20120185245Abstract: An apparatus is provided with a device storing machine readable code and a processor executing the machine readable code. The machine readable code includes sound setting code and audio processing code. The sound setting code detects use of a microphone and sets sound characteristics that are suitable for conversation in response to detecting the use of the microphone. The audio processing code processes sound on the basis of the sound characteristics set by the sound setting code.Type: ApplicationFiled: January 10, 2012Publication date: July 19, 2012Applicant: LENOVO (SINGAPORE) PTE, LTD.Inventors: Shinichi Kikuchi, Hironari Nishino, Yasushi Tsukamoto
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Publication number: 20120179462Abstract: 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.Type: ApplicationFiled: March 21, 2012Publication date: July 12, 2012Inventor: David Klein
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METHOD FOR JOINTLY OPTIMIZING NOISE REDUCTION AND VOICE QUALITY IN A MONO OR MULTI-MICROPHONE SYSTEM
Publication number: 20120179461Abstract: 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.Type: ApplicationFiled: March 19, 2012Publication date: July 12, 2012Inventors: Mark Every, Carlos Avendano -
Publication number: 20120166188Abstract: 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.Type: ApplicationFiled: December 28, 2010Publication date: June 28, 2012Applicant: INTERNATIONAL BUSINESS MACHINES CORPORATIONInventors: Al Chakra, Simon Peter O'Doherty
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Publication number: 20120143604Abstract: 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.Type: ApplicationFiled: December 7, 2010Publication date: June 7, 2012Inventor: Rita Singh
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Publication number: 20120136656Abstract: A method for reducing ringing in a signal output from a filter comprising inputting a signal into a filter; filtering a first portion of the input signal to generate a filtered portion of the output signal; analyzing the filtered portion of the output signal; detecting if ringing is present in the filtered portion of the output signal based on said analysis; and adjusting the filter characteristics to reduce ringing in a subsequent filtered portion of the output signal if it is determined that ringing is present.Type: ApplicationFiled: February 6, 2012Publication date: May 31, 2012Applicant: Skype LimitedInventor: Koen Vos
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Publication number: 20120123774Abstract: An apparatus, electronic apparatus and method for adjusting jitter buffer is provided. A previous jitter buffer size based on a jitter buffer size determined according to an adaptive jitter buffer size calculation algorithm is applied in predicting a jitter buffer size of future time such that the predicted jitter buffer size is applied to obtain a jitter buffer size of a valid time. The audio quality of the speech transmitted over a packet switched network is enhanced.Type: ApplicationFiled: September 23, 2011Publication date: May 17, 2012Applicant: Electronics and Telecommunications Research InstituteInventors: Seung-Han CHOI, Do-Young KIM, Byung-Sun LEE
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Publication number: 20120123770Abstract: 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.Type: ApplicationFiled: September 12, 2011Publication date: May 17, 2012Applicant: Industry-Academic Cooperation Foundation, Yonsei UniversityInventors: Hong Goo Kang, Min Seok Choi, Ho Seon Shin
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Publication number: 20120123772Abstract: 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.Type: ApplicationFiled: November 14, 2011Publication date: May 17, 2012Applicant: Broadcom CorporationInventors: Jes Thyssen, Huaiyu Zeng, Juin-Hwey Chen, Nelson Sollenberger, Xianxian Zhang
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Publication number: 20120123775Abstract: Provided are methods and systems for improving quality of speech communications. The method may be for improving quality of speech communications in a system having a speech encoder configured to encode a first audio signal using a first set of encoding parameters associated with a first noise suppressor. A method may involve receiving a second audio signal at a second noise suppressor which provides much higher quality noise suppression than the first noise suppressor. The second audio signal may be generated by a single microphone or a combination of multiple microphones. The second noise suppressor may suppress the noise in the second audio signal to generate a processed signal which may be sent to a speech encoder. A second set of encoding parameters may be provided by the second noise suppressor for use by the speech encoder when encoding the processed signal into corresponding data.Type: ApplicationFiled: November 14, 2011Publication date: May 17, 2012Inventors: Carlo Murgia, Scott Isabelle
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Publication number: 20120123771Abstract: 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.Type: ApplicationFiled: September 30, 2011Publication date: May 17, 2012Applicant: Broadcom CorporationInventors: Juin-Hwey CHEN, Jes THYSSEN, Xianxian ZHANG, Huaiyu ZENG
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Publication number: 20120116753Abstract: In order to reduce interference in an audio signal during a call on a mobile communication device, a plurality of transforms of the audio signal is performed, each transform containing phase information and amplitude information of corresponding samples of the audio signal. The results of the transforms are then averaged in order to generating a compensation signal that can be subtracted from the audio signal.Type: ApplicationFiled: November 3, 2011Publication date: May 10, 2012Applicant: SONY ERICSSON MOBILE COMMUNICATIONS ABInventors: Jonny STRANDH, Kaj ULLÉN
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Publication number: 20120116754Abstract: Techniques are described herein that suppress noise in a Mel-filtered spectral domain. For example, a window may be applied to a representation of a speech signal in a time domain. The windowed representation in the time domain may be converted to a subsequent representation of the speech signal in the Mel-filtered spectral domain. A noise suppression operation may be performed with respect to the subsequent representation to provide noise-suppressed Mel coefficients.Type: ApplicationFiled: March 22, 2011Publication date: May 10, 2012Applicant: Broadcom CorporationInventor: Jonas Borgstrom
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Publication number: 20120116759Abstract: 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 (QCOD), a bandwidth related distortion parameter (BW) and a presentation level distortion parameter (PL) of a speech signal; extracting a first coefficient (?1) and a second coefficient (?2), the first coefficient and the second coefficient being dependent on the coding distortion parameter; and calculating a signal quality measure (Q), where the signal quality measure is QCOD'?1 Bw+(?2, PL,—using the signal quality measure in a quality estimation of the speech signal.Type: ApplicationFiled: July 26, 2010Publication date: May 10, 2012Inventors: Mats Folkesson, Volodya Grancharov
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Publication number: 20120116758Abstract: 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.Type: ApplicationFiled: November 3, 2011Publication date: May 10, 2012Inventors: Carlo Murgia, Scott Isabelle
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Publication number: 20120101814Abstract: Various techniques are disclosed for improving packet loss concealment to reduce artifacts by using audio character measures of the audio signal. These techniques include attenuation to a noise fill instead of attenuation to silence, varying how long to wait before attenuating the extrapolation, varying the rate of attenuation of the extrapolation, attenuating periodic extrapolation at a different rate than non-periodic extrapolation, and performing period extrapolation on successively longer fill data based on the audio character measures, adjusting weighting between periodic and non-periodic extrapolation based on the audio character measures, and adjusting weighting between periodic extrapolation and non-periodic extrapolation non-linearly.Type: ApplicationFiled: October 25, 2010Publication date: April 26, 2012Applicant: POLYCOM, INC.Inventor: Eric David Elias
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Publication number: 20120101816Abstract: A level of ambient noise at a local device is determined. A dynamic range compression (DRC) gain is computed based on the level of ambient noise at the local device. An additional gain factor is computed. A total gain is computed based on an adding of the DRC gain and the additional gain factor. An amplitude of an audio signal is adjusted based on the total gain, wherein the audio signal was transmitted from a remote device and received by the local device.Type: ApplicationFiled: December 29, 2011Publication date: April 26, 2012Inventor: ADORAM ERELL
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Publication number: 20120095759Abstract: A speech enhancement system that improves the intelligibility and the perceived quality of processed speech includes a frequency transformer and a spectral compressor. The frequency transformer converts speech signals from the time domain to the frequency domain. The spectral compressor compresses a pre-selected portion of the high frequency band and maps the compressed high frequency band to a lower band limited frequency range.Type: ApplicationFiled: December 23, 2011Publication date: April 19, 2012Inventors: Phillip A. Hetherington, Xueman Li
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Publication number: 20120095755Abstract: An audio signal processing system including a time-frequency conversion unit which converts an audio signal in time domain into frequency domain in frame units so as to calculate a frequency spectrum of the audio signal, a spectral change calculation unit which calculates an amount of change between a frequency spectrum of a first frame and a frequency spectrum of a second frame before the first frame based on the frequency spectrum of the first frame and the frequency spectrum of the second frame, and a judgment unit which judges the type of the noise which is included in the audio signal of the first frame in accordance with the amount of spectral change.Type: ApplicationFiled: December 19, 2011Publication date: April 19, 2012Applicant: FUJITSU LIMITEDInventors: Takeshi Otani, Taro Togawa, Masanao Suzuki, Yasuji Ota
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Publication number: 20120084083Abstract: 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.Type: ApplicationFiled: October 4, 2011Publication date: April 5, 2012Applicant: SAMSUNG ELECTRONICS CO., LTD.Inventors: Gun-Hyun YOON, Dong-Won LEE, Ju-Hee CHANG, Koong-Hoon NAM
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Publication number: 20120078620Abstract: 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.Type: ApplicationFiled: December 9, 2011Publication date: March 29, 2012Applicant: QNX Software Systems Co.Inventor: Phillip A. Hetherington
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Publication number: 20120059649Abstract: 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 (107) operates a digital received voice signal with a tap coefficient to generate a pseudo echo; a subtractor (108) subtracts the pseudo echo from a digital transmitted voice signal to generate a residual signal; and an amplitude limiting circuit (110) limits the absolute value of the amplitude of the digital received voice signal to be equal to or smaller than a predetermined threshold which ensures that all of a D/A converter (101), a power amplifier (102), a speaker (103), a microphone (104), a microphone amplifier (105), and an A/D converter (106) operate in a linear operation area, and outputs the amplitude-limited digital received voice signal to the D/A converter (101) and the adaptive filter (107).Type: ApplicationFiled: March 19, 2010Publication date: March 8, 2012Applicant: YUGENGAISYA CEPSTRUMInventor: Akio Yamaguchi
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Publication number: 20120059648Abstract: 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.Type: ApplicationFiled: February 28, 2011Publication date: March 8, 2012Inventors: Gregory C. BURNETT, Eric F. BREITFELLER
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Publication number: 20120059650Abstract: 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.Type: ApplicationFiled: April 12, 2010Publication date: March 8, 2012Applicant: FRANCE TELECOMInventors: Julien Faure, Adrien Leman
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Publication number: 20120046943Abstract: An apparatus and a method for voice communication of a mobile terminal are provided. More particularly, an apparatus and a method for clearly receiving a counterpart user's voice signal in a mobile terminal positioned at a place where a noise occurs are provided. The apparatus includes an input unit, an extension signal generator, and an adder. The input unit receives a voice signal. The extension signal generator generates, based on a voice signal received via the input unit, a harmonics signal corresponding to a frequency band that represents a reaction sensitive to a sense of hearing. The adder merges the generated harmonics signal with the received voice signal.Type: ApplicationFiled: August 17, 2011Publication date: February 23, 2012Applicant: SAMSUNG ELECTRONICS CO. LTD.Inventors: Nam-Woog LEE, Jae-Hyun KIM, Sang-Jin KIM, Baek-Kwon SON
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Publication number: 20120045074Abstract: Disclosed herein are system, method and apparatus with environmental noise cancellation. The instant disclosure is particularly adapted to a receiver module having at least two inputs. The two inputs respectively receive a main audio portion and the audio with majority of environmental noise. The system firstly calibrates the audio signals to reduce the error caused by the difference between the two inputs. An adaptive beamforming technology and a speech extractor are respectively used to extract the environmental noise portion with less main audio and the main audio portion with less noise. After a process of time-to-frequency domain transformation, a non-linear noise suppression technology is introduced into estimating the environmental noise and acquiring a gain. After noise suppression processed with the gain, a sequence of audio signals is output after a frequency-to-time domain transformation.Type: ApplicationFiled: January 7, 2011Publication date: February 23, 2012Applicant: C-MEDIA ELECTRONICS INC.Inventors: YUEPENG LI, FENGHAI QIU, HUA GAO
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Publication number: 20120035921Abstract: 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.Type: ApplicationFiled: August 25, 2011Publication date: February 9, 2012Applicant: QNX Software Systems Co.Inventors: Xueman Li, Rajeev Nongpiur, Phillip A. Hetherington