Silence Decision Patents (Class 704/215)
  • Patent number: 10381014
    Abstract: A comfort noise controller for generating CN (Comfort Noise) control parameters is described. A buffer of a predetermined size is configured to store CN parameters for SID (Silence Insertion Descriptor) frames and active hangover frames. A subset selector is configured to determine a CN parameter subset relevant for SID frames based on the age of the stored CN parameters and on residual energies. A comfort noise control parameter extractor (50B) is configured to use the determined CN parameter subset to determine the CN control parameters for a first SID frame following an active signal frame.
    Type: Grant
    Filed: August 22, 2017
    Date of Patent: August 13, 2019
    Assignee: TELEFONAKTIEBOLAGET LM ERICSSON (PUBL)
    Inventor: Tomas Jansson Toftgård
  • Patent number: 10311868
    Abstract: A device performs a method for using image data to aid voice recognition. The method includes the device capturing (302) image data of a vicinity of the device and adjusting (304), based on the image data, a set of parameters for voice recognition performed by the device (102). The set of parameters for the device performing voice recognition include, but are not limited to: a trigger threshold of a trigger for voice recognition; a set of beamforming parameters; a database for voice recognition; and/or an algorithm for voice recognition. The algorithm may include using noise suppression or using acoustic beamforming.
    Type: Grant
    Filed: March 21, 2017
    Date of Patent: June 4, 2019
    Assignee: Google Technology Holdings LLC
    Inventors: Robert A. Zurek, Adrian M. Schuster, Fu-Lin Shau, Jincheng Wu
  • Patent number: 10269375
    Abstract: The disclosed embodiments illustrate a method for classifying one or more audio segments of an audio signal. The method includes determining one or more first features of a first audio segment of the one or more audio segments. The method further includes determining one or more second features based on the one or more first features. The method includes determining one or more third features of the first audio segment, wherein each of the one or more third features is determined based on a second feature of the one or more second features of the first audio segment and at least one second feature associated with a second audio segment. Additionally, the method includes classifying the first audio segment either in an interrogative category or a non-interrogative category based on one or more of the one or more second features and the one or more third features.
    Type: Grant
    Filed: April 22, 2016
    Date of Patent: April 23, 2019
    Assignee: CONDUENT BUSINESS SERVICES, LLC
    Inventors: Harish Arsikere, Arunasish Sen, Prathosh Aragulla Prasad
  • Patent number: 10248917
    Abstract: Systems and methods for developing and utilizing a contactability profile are disclosed. Contact information may be managed by receiving activity data associated with an entity, processing the received activity data, generating at least one contactability profile associated with the entity based upon a result of the processing, and storing the generated contactability profile.
    Type: Grant
    Filed: October 14, 2005
    Date of Patent: April 2, 2019
    Assignee: CAPITAL ONE SERVICES, LLC
    Inventors: Anthony Paul Reynolds, Daniel J. Welker
  • Patent number: 10237647
    Abstract: A beamformer system that can isolate a desired portion of an audio signal resulting from a microphone array. A combination of beamformers is used to dampen undesired noise, whether diffuse or coherent. A fixed beamformer is used to dampen diffuse noise while an adaptive beamformer is used to cancel directional coherent noise. The adaptive beamformer isolates and weights audio from various directions. The weights may vary depending on the isolated desired audio signal, dynamically adjusting the step-size adjustments to the weights.
    Type: Grant
    Filed: March 1, 2017
    Date of Patent: March 19, 2019
    Assignee: Amazon Technologies, Inc.
    Inventor: Amit Singh Chhetri
  • Patent number: 10157359
    Abstract: A method for using oral feedback to facilitate user selection among a plurality of service devices. The method comprises a plurality of service devices. Each service device has a dynamically-updatable quality control profile. A first service device receives a request from the first user to perform a first service job. The first service device performs the first service job. A first recording device located proximate to the first service device obtains oral feedback from the first user regarding the quality of the first service job. The first service device analyzes the oral feedback using natural language processing to form an analysis. The first service device modifies the first quality control profile of the first service device based on the analysis.
    Type: Grant
    Filed: February 15, 2017
    Date of Patent: December 18, 2018
    Assignee: International Business Machines Corporation
    Inventors: Joshi M. Chirayil, Rajendra S. Kanyal, Dhaval K. Shah, Anupam Varma
  • Patent number: 10089993
    Abstract: An apparatus for encoding audio information is provided. The apparatus for encoding audio information includes a selector for selecting a comfort noise generation mode from two or more comfort noise generation modes depending on a background noise characteristic of an audio input signal, and an encoding unit for encoding the audio information, wherein the audio information includes mode information indicating the selected comfort noise generation mode.
    Type: Grant
    Filed: January 27, 2017
    Date of Patent: October 2, 2018
    Assignee: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
    Inventors: Emmanuel Ravelli, Martin Dietz, Wolfgang Jaegers, Christian Neukam, Stefan Reuschl
  • Patent number: 9953272
    Abstract: A machine learning system for evaluating at least one characteristic of a heart valve, an inflow tract, an outflow tract or a combination thereof may include a training mode and a production mode. The training mode may be configured to train a computer and construct a transformation function to predict an unknown anatomical characteristic and/or an unknown physiological characteristic of a heart valve, inflow tract and/or outflow tract, using a known anatomical characteristic and/or a known physiological characteristic the heart valve, inflow tract and/or outflow tract. The production mode may be configured to use the transformation function to predict the unknown anatomical characteristic and/or the unknown physiological characteristic of the heart valve, inflow tract and/or outflow tract, based on the known anatomical characteristic and/or the known physiological characteristic of the heart valve, inflow tract and/or outflow tract.
    Type: Grant
    Filed: April 26, 2016
    Date of Patent: April 24, 2018
    Assignee: Stenomics, Inc.
    Inventor: Michael A. Singer
  • Patent number: 9906795
    Abstract: When removing a block distortion occurring in a local decoded image, a loop filtering part 11 of an image coding device carries out a filtering process on each of signal components (a luminance signal component and color difference signal components) after setting the intensity of a filter for removing the block distortion for each of the signal components according to a coding mode (an intra coding mode or an inter coding mode) selected by a coding controlling part 1.
    Type: Grant
    Filed: April 5, 2017
    Date of Patent: February 27, 2018
    Assignee: MITSUBISHI ELECTRIC CORPORATION
    Inventors: Shunichi Sekiguchi, Kazuo Sugimoto, Hiroharu Sakate, Tokumichi Murakami, Akira Minezawa
  • Patent number: 9866972
    Abstract: Described herein is a MEMS acoustic transducer device provided with a micromechanical detection structure that detects acoustic-pressure waves and supplies a transduced electrical quantity, and with an integrated circuit operatively coupled to the micromechanical detection structure and having a reading module that generates at output an audio signal as a function of the transduced electrical quantity. The integrated circuit is further provided with a recognition module, which recognizes a sound activity event associated to the transduced electrical quantity. The MEMS acoustic transducer has an output that supplies at output a data signal that carries information regarding recognition of the sound activity event.
    Type: Grant
    Filed: November 4, 2014
    Date of Patent: January 9, 2018
    Assignee: STMICROELECTRONICS S.R.L.
    Inventors: Marco Veneri, Alessandro Morcelli
  • Patent number: 9830925
    Abstract: An automatic speech recognition engine and a method of using the engine is described. The method pertains to front-end processing an audio signal and includes the steps of: identifying a plurality of voiced-frames of the audio signal; determining that one or more of the plurality of voiced-frames have a signal-to-noise (SNR) value greater than a first predetermined threshold; and based on the determination, bypassing noise suppression for the one or more of the plurality of voiced-frames.
    Type: Grant
    Filed: October 22, 2014
    Date of Patent: November 28, 2017
    Assignee: GM Global Technology Operations LLC
    Inventors: Gaurav Talwar, Xufang Zhao, III, Robert D. Sims, III, Md Foezur Rahman Chowdhury
  • Patent number: 9773505
    Abstract: An encoding apparatus and a decoding apparatus in a transform between a Modified Discrete Cosine Transform (MDCT)-based coder and a different coder are provided. The encoding apparatus may encode additional information to restore an input signal encoded according to the MDCT-based coding scheme, when switching occurs between the MDCT-based coder and the different coder. Accordingly, an unnecessary bitstream may be prevented from being generated, and minimum additional information may be encoded.
    Type: Grant
    Filed: September 18, 2009
    Date of Patent: September 26, 2017
    Assignees: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE, KWANGWOON UNIVERSITY INDUSTRY-ACADEMIC COLLABORATION FOUNDATION
    Inventors: Seung Kwon Beack, Tae Jin Lee, Min Je Kim, Dae Young Jang, Kyeongok Kang, Jin Woo Hong, Ho Chong Park, Young-cheol Park
  • Patent number: 9775110
    Abstract: Methods and apparatus for reducing power consumption in a wireless device operating in a discontinuous transmission (DTX) mode while using a voice over Long Term Evolution (VoLTE) service. The wireless device detects a period of voice inactivity and transmits one or more silence descriptor (SID_UPDATE) frames to a second wireless device in place of encoded speech frames. The SID_UPDATE frames are transmitted periodically based on measurements of comfort noise parameters. The wireless device determines a difference between weighted averages of comfort noise (CN) parameters of two sequences of encoded speech frames. When the difference exceeds a difference threshold, a SID_UPDATE frame is transmitted. Additionally, in some embodiments, a SID_UPDATE frame is transmitted when the weighted average of CN parameters exceeds a parameter threshold and/or when a time between SID_UPDATE frames or time elapsed after entering a silence state exceeds one or more time thresholds.
    Type: Grant
    Filed: September 25, 2014
    Date of Patent: September 26, 2017
    Assignee: Apple Inc.
    Inventors: Sreevalsan Vallath, Carola Faronius
  • Patent number: 9734835
    Abstract: A voice decoding apparatus includes an MBE-type decoder, a sampling convertor, a non-linear components generator and an adder. The decoder decodes digital voice-encoded information to generate a first decoded voice signal. The convertor converts the first decoded voice signal to a second decoded voice signal with a higher sampling frequency. The generator performs a non-linear process to the first or second decoded voice signal to generate an additional voice signal with the same sampling frequency as the second decoded voice signal. The additional voice signal has components in a frequency band in which the first decoded voice signal has no component and continuing to another frequency band of the first decoded voice signal. The adder adds the second decoded voice signal to the additional voice signal.
    Type: Grant
    Filed: February 5, 2015
    Date of Patent: August 15, 2017
    Assignee: Oki Electric Industry Co., Ltd.
    Inventor: Masaru Fujieda
  • Patent number: 9715883
    Abstract: In an embodiment, bitstream elements of sub-frames are encoded differentially to a global gain value so that a change of the global gain value results in an adjustment of an output level of the decoded representation of the audio content. Concurrently, the differential coding saves bits. Even further, the differential coding enables the lowering of the burden of globally adjusting the gain of an encoded bitstream. In another embodiment, a global gain control across CELP coded frames and transform coded frames is achieved by co-controlling the gain of the codebook excitation of the CELP codec, along with a level of the transform or inverse transform of the transform coded frames. In another embodiment, the gain value determination in CELP coding is performed in the weighted domain of the excitation signal.
    Type: Grant
    Filed: May 12, 2016
    Date of Patent: July 25, 2017
    Assignee: Fraundhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
    Inventors: Ralf Geiger, Guillaume Fuchs, Markus Multrus, Bernhard Grill
  • Patent number: 9711157
    Abstract: An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal includes a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also includes an audio stream provider for providing the audio stream such that the audio stream includes information describing an audio content of the frequency bands and information describing the multi-band quantization error. A decoder for providing a decoded representation of an audio signal on the basis of an encoded audio stream representing spectral components of frequency bands of the audio signal includes a noise filler for introducing noise into spectral components of a plurality of frequency bands to which separate frequency band gain information is associated on the basis of a common multi-band noise intensity value.
    Type: Grant
    Filed: December 24, 2014
    Date of Patent: July 18, 2017
    Assignee: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
    Inventors: Nikolaus Rettelbach, Bernhard Grill, Guillaume Fuchs, Stefan Geyersberger, Markus Multrus, Harald Popp, Juergen Herre, Stefan Wabnik, Gerald Schuller, Jens Hirschfeld
  • Patent number: 9614488
    Abstract: An information processing apparatus includes an audio processing unit that performs a predetermined process for an input audio signal, and a power saving control unit that allows the audio processing unit to transition into a power saving state, depending on the length of a silence section in the audio signal.
    Type: Grant
    Filed: March 20, 2015
    Date of Patent: April 4, 2017
    Assignee: SONY CORPORATION
    Inventors: Takashi Kobayashi, Tomohiro Katori
  • Patent number: 9515748
    Abstract: A decoding apparatus includes a sound code input unit that receives a sound code output from an encoding apparatus through a sound wave reception device; a frame division unit that divides the sound code depending on a predetermined time interval to generate a plurality of frames; a frequency identification unit that identifies a frequency corresponding to each of the plurality of the frames through frequency analysis for each of the plurality of the frames; and an information generation unit that determines a frequency band, to which each of the identified frequencies corresponds, from an audible sound wave frequency band and a non-audible sound wave frequency band, and a plurality of partial information based on the frequency band and each of the identified frequencies, and generates information corresponding to the sound code based on the plurality of the partial information.
    Type: Grant
    Filed: September 24, 2014
    Date of Patent: December 6, 2016
    Assignee: POWERVOICE CO., LTD.
    Inventors: Hee Suk Jeong, Se Hun Chin, Hyung Yup Lee, Jong Sang Tack
  • Patent number: 9454959
    Abstract: Speech recognition systems often process speech by employing models and analyzing audio data. An embodiment of the method and corresponding system described herein allow for passive monitoring of, for example, conversation between user(s) to determine context to use to prime model(s) for later speech recognition requests submitted to the speech recognition system. The embodiment improves the results of the speech recognition system by updating speech recognition model(s) with contextual information of the conversation. This increases the probability that the speech recognition system interprets the conversation to contextually relevant information.
    Type: Grant
    Filed: November 2, 2012
    Date of Patent: September 27, 2016
    Assignee: Nuance Communications, Inc.
    Inventors: Nils Lenke, William F. Ganong, III
  • Patent number: 9449609
    Abstract: Acoustic noise in an audio signal is reduced by calculating a speech probability presence (SPP) factor using minimum mean square error (MMSE). The SPP factor, which has a value typically ranging between zero and one, is modified or warped responsive to a value obtained from the evaluation of a sigmoid function, the shape of which is determined by a signal-to-noise ratio (SNR), which is obtained by an evaluation of the signal energy and noise energy output from a microphone over time. The shape and aggressiveness of the sigmoid function is determined using an extrinsically-determined SNR, not determined by the MMSE determination. The extrinsically-determined SNR is obtained from a long term history of previously-determined speech presence probabilities and a long term history of previously-determined noise histories.
    Type: Grant
    Filed: November 7, 2013
    Date of Patent: September 20, 2016
    Assignee: Continental Automotive Systems, Inc.
    Inventors: Guillaume Lamy, Bijal Joshi
  • Patent number: 9424854
    Abstract: A method for processing audio data includes determining a first common scalefactor value for representing quantized audio data in a frame. A second common scalefactor value is determined for representing the quantized audio data in the frame. A line equation common scalefactor value is determined from the first and second common scalefactor values.
    Type: Grant
    Filed: October 7, 2013
    Date of Patent: August 23, 2016
    Assignee: Intel Corporation
    Inventors: Dmitry N. Budnikov, Igor Igor Chikalov, Sergey N. Zheltov
  • Patent number: 9424531
    Abstract: A machine learning system for evaluating at least one characteristic of a heart valve, an inflow tract, an outflow tract or a combination thereof may include a training mode and a production mode. The training mode may be configured to train a computer and construct a transformation function to predict an unknown anatomical characteristic and/or an unknown physiological characteristic of a heart valve, inflow tract and/or outflow tract, using a known anatomical characteristic and/or a known physiological characteristic the heart valve, inflow tract and/or outflow tract. The production mode may be configured to use the transformation function to predict the unknown anatomical characteristic and/or the unknown physiological characteristic of the heart valve, inflow tract and/or outflow tract, based on the known anatomical characteristic and/or the known physiological characteristic of the heart valve, inflow tract and/or outflow tract.
    Type: Grant
    Filed: April 7, 2015
    Date of Patent: August 23, 2016
    Assignee: STENOMICS, INC.
    Inventor: Michael A. Singer
  • Patent number: 9031619
    Abstract: A method includes, during operation of a mobile handset when a voice call is in progress with a remote party, detecting an occurrence of a start of a discontinuous transmission period that results from a cessation of speech by the remote party; and activating a visual indicator of the mobile handset to assume a first state that indicates that a speech signal is not being received from the remote party.
    Type: Grant
    Filed: September 30, 2010
    Date of Patent: May 12, 2015
    Assignee: Nokia Corporation
    Inventor: Francis Quiers
  • Patent number: 9020816
    Abstract: A method, system and apparatus are shown for identifying non-language speech sounds in a speech or audio signal. An audio signal is segmented and feature vectors are extracted from the segments of the audio signal. The segment is classified using a hidden Markov model (HMM) that has been trained on sequences of these feature vectors. Post-processing components can be utilized to enhance classification. An embodiment is described in which the hidden Markov model is used to classify a segment as a language speech sound or one of a variety of non-language speech sounds. Another embodiment is described in which the hidden Markov model is trained using discriminative learning.
    Type: Grant
    Filed: August 13, 2009
    Date of Patent: April 28, 2015
    Assignee: 21CT, Inc.
    Inventor: Matthew McClain
  • Patent number: 9009034
    Abstract: A Voice Activity Detection/Silence Suppression (VAD/SS) system is connected to a channel of a transmission pipe. The channel provides a pathway for the transmission of energy. A method for operating a VAD/SS system includes detecting the energy on the channel, and activating or suppressing activation of the VAD/SS system depending upon the nature of the energy detected on the channel.
    Type: Grant
    Filed: November 12, 2014
    Date of Patent: April 14, 2015
    Assignee: AT&T Intellectual Property II, L.P.
    Inventors: Bing Chen, James H. James
  • Patent number: 8983851
    Abstract: A noise filler for providing a noise-filled spectral representation of an audio signal on the basis of an input spectral representation of the audio signal has a spectral region identifier configured to identify spectral regions of the input spectral representation spaced from non-zero spectral regions of the input spectral representation by at least one intermediate spectral region, to obtain identified spectral regions, and a noise inserter configured to selectively introduce noise into the identified spectral regions to obtain the noise-filled spectral representation of the audio signal. A noise filling parameter calculator for providing a noise filling parameter on the basis of a quantized spectral representation of an audio signal has a spectral region identifier, as mentioned above, and a noise value calculator configured to selectively consider quantization errors of the identified spectral regions for a calculation of the noise filling parameter.
    Type: Grant
    Filed: January 11, 2011
    Date of Patent: March 17, 2015
    Assignee: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
    Inventors: Nikolaus Rettelbach, Bernhard Grill, Guillaume Fuchs, Stefan Geyersberger, Markus Multrus, Harald Popp, Juergen Herre, Stefan Wabnik, Gerald Schuller, Jens Hirschfeld
  • Patent number: 8977556
    Abstract: Embodiments of the present invention relate to a voice detector receiving an input signal that is divided into sub-signals that represent a frequency sub-band. The voice detector calculates, for each sub-band, a signal-to-noise (SNR) value based on a corresponding sub-signal for each sub-band and a background signal for each sub-band. The voice detector also calculates a power SNR value for each sub-band, where at least one of the power SNR values is calculated based on a non-linear function. The voice detector forms a single value based on the calculated power SNR values and compares the single value and a given threshold value to make a voice activity decision presented on an output port.
    Type: Grant
    Filed: March 26, 2012
    Date of Patent: March 10, 2015
    Assignee: Telefonaktiebolaget LM Ericsson (Publ)
    Inventor: Martin Sehlstedt
  • Patent number: 8954324
    Abstract: Voice activity detection using multiple microphones can be based on a relationship between an energy at each of a speech reference microphone and a noise reference microphone. The energy output from each of the speech reference microphone and the noise reference microphone can be determined. A speech to noise energy ratio can be determined and compared to a predetermined voice activity threshold. In another embodiment, the absolute value of the autocorrelation of the speech and noise reference signals are determined and a ratio based on autocorrelation values is determined. Ratios that exceed the predetermined threshold can indicate the presence of a voice signal. The speech and noise energies or autocorrelations can be determined using a weighted average or over a discrete frame size.
    Type: Grant
    Filed: September 28, 2007
    Date of Patent: February 10, 2015
    Assignee: QUALCOMM Incorporated
    Inventors: Song Wang, Samir Kumar Gupta, Eddie L. T. Choy
  • Patent number: 8942987
    Abstract: A clear picture of who is speaking in a setting where there are multiple input sources (e.g., a conference room with multiple microphones) can be obtained by comparing input channels against each other. The data from each channel can not only be compared, but can also be organized into portions which logically correspond to statements by a user. These statements, along with information regarding who is speaking, can be presented in a user friendly format via an interactive timeline which can be updated in real time as new audio input data is received.
    Type: Grant
    Filed: March 21, 2014
    Date of Patent: January 27, 2015
    Assignee: Jefferson Audio Video Systems, Inc.
    Inventors: Matthew David Bader, Nathan David Cole
  • Patent number: 8938389
    Abstract: A frame extracting means 71 extracts frames from sample data as voice data in which whether each frame is an active voice frame or a non-active voice frame is already known. A feature quantity calculating means 72 calculates multiple feature quantities of each of the frames. A feature quantity integrating means 73 calculates an integrated feature quantity of the multiple feature quantities. A judgment means 74 judges whether each of the frames is an active voice frame or a non-active voice frame. An erroneous feature quantity calculation value calculating means 75 obtains a first erroneous feature quantity calculation value and a second erroneous feature quantity calculation value by executing prescribed calculations. A weight updating means 76 updates weights used for weighting so that the rate between the first erroneous feature quantity calculation value and the second erroneous feature quantity calculation value approaches a prescribed value.
    Type: Grant
    Filed: December 7, 2009
    Date of Patent: January 20, 2015
    Assignee: NEC Corporation
    Inventors: Takayuki Arakawa, Masanori Tsujikawa
  • Patent number: 8930184
    Abstract: A signal bandwidth extending apparatus including: a bandwidth extending section configured to extend a frequency bandwidth of a target signal, the target signal included in an input signal; a calculating section configured to calculate a degree of the target signal included in the input signal; and a controller configured to change a method of extending the frequency bandwidth by the bandwidth extending section according to a result of the calculating section.
    Type: Grant
    Filed: September 14, 2009
    Date of Patent: January 6, 2015
    Assignee: Kabushiki Kaisha Toshiba
    Inventors: Takashi Sudo, Masataka Osada
  • Patent number: 8924200
    Abstract: A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.
    Type: Grant
    Filed: September 28, 2011
    Date of Patent: December 30, 2014
    Assignee: Motorola Mobility LLC
    Inventors: Jonathan A. Gibbs, James P. Ashley, Udar Mittal
  • Patent number: 8924207
    Abstract: A method and apparatus for transcoding audio data. The method includes determining if AAC joint stereo exists, running a reference AC-3 rematrixing when the AAC joint stereo does not exist, when AAC joint stereo does exist, enabling rematrixing when the number of corresponding AAC bands is greater than half the size of the band, otherwise, running reference AC-3 rematrixing.
    Type: Grant
    Filed: July 20, 2010
    Date of Patent: December 30, 2014
    Assignee: Texas Instruments Incorporated
    Inventor: Mohamed Farouk Mansour
  • Patent number: 8913512
    Abstract: A telecommunication apparatus (100, 200) enabled for high-speed packet access is disclosed. The apparatus (100, 200) is arranged to operate according to a reduced and a further reduced mode of transmission of dedicated physical control channel transmission, and having a data transmission controller (102, 202) arranged to control sporadic data transmissions. The data transmission controller (102, 202) is arranged to determine if omission of a sporadic data transmission will significantly degrade performance, and if not, disable transmission of that data transmission. A method of controlling sporadic data transmissions for such an apparatus is also disclosed, as well as a computer program for implementing the method.
    Type: Grant
    Filed: October 16, 2008
    Date of Patent: December 16, 2014
    Assignee: Telefonaktiebolaget LM Ericsson (publ)
    Inventors: Hans Hannu, Jan Christoffersson, Min Wang
  • Patent number: 8909524
    Abstract: Embodiments of the present invention provide an adaptive noise canceling system. The adaptive noise canceling system may be used in a handset to cancel background noise by generating an anti-noise signal. The adaptive noise canceling system may include first input to receive a first signal from a feedforward microphone; a second input to receive a second signal from an error microphone; a controller coupled to the inputs, the controller configured to adaptively generate an anti-noise signal according to the received signals, wherein the controller derives a profile of the anti-noise signal from the first signal and derives a magnitude of the anti-noise signal from both first and second signal; and an output to transmit the anti-noise signal to a speaker.
    Type: Grant
    Filed: June 7, 2011
    Date of Patent: December 9, 2014
    Assignee: Analog Devices, Inc.
    Inventors: Thomas Stoltz, Kim Spetzler Berthelsen, Robert Adams
  • Patent number: 8909519
    Abstract: A Voice Activity Detection/Silence Suppression (VAD/SS) system is connected to a channel of a transmission pipe. The channel provides a pathway for the transmission of energy. A method for operating a VAD/SS system includes detecting the energy on the channel, and activating or suppressing activation of the VAD/SS system depending upon the nature of the energy detected on the channel.
    Type: Grant
    Filed: March 10, 2014
    Date of Patent: December 9, 2014
    Assignee: AT&T Intellectual Property II, L.P.
    Inventors: Bing Chen, James H. James
  • Patent number: 8892229
    Abstract: An audio apparatus according to an embodiment includes an audio signal receiving unit, a music gap signal receiving unit, a playback unit, and a determining unit. The audio signal receiving unit receives an audio signal in which successive multiple music data are contained in a single block of data. The determining unit determines a boundary of the music data on the basis of the time at which the music gap signal that indicates the boundary of the music data by the music gap signal receiving unit and the duration of a silent period in the audio signal that is played back by the playback unit.
    Type: Grant
    Filed: May 14, 2012
    Date of Patent: November 18, 2014
    Assignee: Fujitsu Ten Limited
    Inventors: Osamu Yasutake, Fumitake Nakamura, Nobutaka Miyauchi, Masanobu Maeda, Masahiko Kubo, Nahoko Kawamura, Machiko Matsui, Hideto Saitoh, Hiroyuki Kubota, Masayuki Takaoka, Masanobu Washio, Yutaka Nishioka
  • Patent number: 8886529
    Abstract: 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: Grant
    Filed: April 12, 2010
    Date of Patent: November 11, 2014
    Assignee: France Telecom
    Inventors: Julien Faure, Adrien Leman
  • Patent number: 8886527
    Abstract: A purpose is to suppress recognition process delay generated due to load in signal processing. Included is a speech input means 10 that inputs a speech signal, an output evaluation means 20 that evaluates whether or not the speech signal input by the speech input means 10 is the speech signal in a sound section, which is a speech section assuming that a speaker is speaking, and outputs the speech signal as a speech signal to be processed only when evaluated as the speech signal in the sound section, a signal processing means 30 that performs signal processing to the speech signal, which is output by the output evaluation means 20 as the speech signal to be processed, and a speech recognition processing means 40 that performs a speech recognition process to the speech signal which is signal-processed by the signal processing means 30.
    Type: Grant
    Filed: April 16, 2009
    Date of Patent: November 11, 2014
    Assignee: NEC Corporation
    Inventor: Toru Iwasawa
  • Patent number: 8868432
    Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.
    Type: Grant
    Filed: September 28, 2011
    Date of Patent: October 21, 2014
    Assignee: Motorola Mobility LLC
    Inventors: Jonathan A. Gibbs, James P. Ashley, Udar Mittal
  • Patent number: 8856001
    Abstract: A speech sound detection apparatus receives an input audio signal (as a sound reception unit), and computes input power that indicates a magnitude of the sound represented by the audio signal (as an input power computation unit). The apparatus estimates a correction function that is a continuous function defining a relation between a certain frequency and a correction coefficient used to approximate the input power computed at that frequency to the reference power predetermined for that frequency (as a correction function estimation unit). The apparatus corrects the input power at every frequency, based upon the correction coefficient that is obtained in accordance with the relation defined by the estimated correction function (as an input power correcting unit). The apparatus further determines whether or not the sound represented by the received audio signal is speech sound, based upon the corrected input power (as a speech sound detection unit).
    Type: Grant
    Filed: September 3, 2009
    Date of Patent: October 7, 2014
    Assignee: NEC Corporation
    Inventors: Tadashi Emori, Masanori Tsujikawa
  • Patent number: 8825478
    Abstract: Audio content is converted to text using speech recognition software. The text is then associated with a distinct voice or a generic placeholder label if no distinction can be made. From the text and voice information, a word cloud is generated based on key words and key speakers. A visualization of the cloud displays as it is being created. Words grow in size in relation to their dominance. When it is determined that the predominant words or speakers have changed, the word cloud is complete. That word cloud continues to be displayed statically and a new word cloud display begins based upon a new set of predominant words or a new predominant speaker or set of speakers. This process may continue until the meeting is concluded. At the end of the meeting, the completed visualization may be saved to a storage device, sent to selected individuals, removed, or any combination of the preceding.
    Type: Grant
    Filed: January 10, 2011
    Date of Patent: September 2, 2014
    Assignee: Nuance Communications, Inc.
    Inventors: Susan Marie Cox, Janani Janakiraman, Fang Lu, Loulwa F Salem
  • Patent number: 8818811
    Abstract: This application relates to a voice activity detection (VAD) apparatus configured to provide a voice activity detection decision for an input audio signal. The VAD apparatus includes a state detector and a voice activity calculator. The state detector is configured to determine, based on the input audio signal, a current working state of the VAD apparatus among at least two different working states. Each of the at least two different working states is associated with a corresponding working state parameter decision set which includes at least one voice activity decision parameter. The voice activity calculator is configured to calculate a voice activity detection parameter value for the at least one voice activity decision parameter of the working state parameter decision set associated with the current working state, and to provide the voice activity detection decision by comparing the calculated voice activity detection parameter value with a threshold.
    Type: Grant
    Filed: June 24, 2013
    Date of Patent: August 26, 2014
    Assignee: Huawei Technologies Co., Ltd
    Inventor: Zhe Wang
  • Patent number: 8781821
    Abstract: A method is disclosed for controlling a voice-activated device by interpreting a spoken command as a series of voiced and non-voiced intervals. A responsive action is then performed according to the number of voiced intervals in the command. The method is well-suited to applications having a small number of specific voice-activated response functions. Applications using the inventive method offer numerous advantages over traditional speech recognition systems including speaker universality, language independence, no training or calibration needed, implementation with simple microcontrollers, and extremely low cost. For time-critical applications such as pulsers and measurement devices, where fast reaction is crucial to catch a transient event, the method provides near-instantaneous command response, yet versatile voice control.
    Type: Grant
    Filed: April 30, 2012
    Date of Patent: July 15, 2014
    Assignee: Zanavox
    Inventor: David Edward Newman
  • Patent number: 8775168
    Abstract: A Yule-Walker based, low-complexity voice activity detector (VAD) is disclosed. An input signal is typically noisy speech (i.e., corrupted with, for example, babble noise). In one embodiment, a first initialization stage of the VAD computes an occurrence of a silent period within the input signal and the AR parameters. The VAD could accordingly compute a tentative adaptive threshold and output hypothesis H1 (which means speech is present) during this stage. During the second initialization stage, the VAD generally builds a database of associated values and computes the adaptive threshold accordingly. The second initialization stage could also output tentative VAD decisions based on the tentative threshold computed in the first initialization stage. Finally, the VAD periodically retrains or updates AR parameters, threshold values and/or the database and outputs VAD decisions accordingly.
    Type: Grant
    Filed: August 3, 2007
    Date of Patent: July 8, 2014
    Assignee: STMicroelectronics Asia Pacific PTE, Ltd.
    Inventors: Karthik Muralidhar, Anoop Kumar Krishna
  • Patent number: 8762158
    Abstract: A method and apparatus for generating synthesis audio signals are provided. The method includes decoding a bitstream; splitting the decoded bitstream into n sub-band signals; generating n transformed sub-band signals by transforming the n sub-band signals in a frequency domain; and generating synthesis audio signals by respectively multiplying the n transformed sub-band signals by values corresponding to synthesis filter bank coefficients.
    Type: Grant
    Filed: August 5, 2011
    Date of Patent: June 24, 2014
    Assignee: Samsung Electronics Co., Ltd.
    Inventors: Hyun-wook Kim, Han-gil Moon, Sang-hoon Lee
  • Patent number: 8762144
    Abstract: A method and apparatus for detecting voice activity are disclosed. The method of detecting voice activity includes: extracting a feature parameter from a frame signal; determining whether the frame signal is a voice signal or a noise signal by comparing the feature parameter with model parameters of a plurality of comparison signals, respectively; and outputting the frame signal when the frame signal is determined to be a voice signal. The apparatus includes a classifier module which extracts a feature parameter from a frame signal, and generating labeling information with respect to the frame signal by comparing the feature parameter with model parameters of a plurality of comparison signals; and a voice detection unit which determines whether the frame signal is a noise signal or a voice signal with reference to the labeling information, and outputting the frame signal when the frame signal is determined to be a voice signal.
    Type: Grant
    Filed: May 3, 2011
    Date of Patent: June 24, 2014
    Assignee: Samsung Electronics Co., Ltd.
    Inventors: Nam-gook Cho, Eun-kyoung Kim
  • Patent number: 8751223
    Abstract: In one implementation, a first voice stream for a packet-switched call is received from a calling party. The first voice stream conforms to a first silence suppression scheme and comprises a plurality of encoded packets for the packet-switched call. A subset of encoded packets are selected from the plurality of encoded packets to create a second voice stream that conforms to a second silence suppression scheme. The second voice stream comprises the subset of encoded packets. The first silence suppression scheme is distinct from the second silence suppression scheme. The second voice stream is forwarded toward a called party for the packet-switched call.
    Type: Grant
    Filed: May 24, 2011
    Date of Patent: June 10, 2014
    Assignee: Alcatel Lucent
    Inventors: Jeffrey A. Hiltner, Alan H. Matten, Dale R. Schumacher, Albert J. Such
  • Patent number: 8731914
    Abstract: A system and method for locating a preferable playback start location after a winding or rewinding action in an audio playing device. In response to an adjustment of the playing location for audio content to a desired playing position, the system determines whether at least one non-speech or silent period of at least a predetermined duration exists within the vicinity of the desired playing position. If at least one such non-speech or silent period exists within the vicinity of the desired playing position, the system adjusts the playing position to fall within one of the at least one non-speech period or silent period.
    Type: Grant
    Filed: November 15, 2005
    Date of Patent: May 20, 2014
    Assignee: Nokia Corporation
    Inventors: Janne Vainio, Hannu J. Mikkola, Jari M. Makinen
  • Patent number: RE46388
    Abstract: There is provided an audio coding device which appropriately sets the quantization bit number by a small calculation amount in each stage when coding an input audio signal by performing multi-stage normalization/quantization. A quantization information calculation section determines total quantization information idwl0, based on normalization information idsf, and allocates the total quantization information idwl0 for quantization information idwl1 and quantization information idwl2. At this time, the quantization information calculation section limits the quantization information idwl1 by a limiter lim1, and allocates the total quantization information idwl0 for quantization information idwl1. If the quantization information idwl1 exceeds the limiter lim1, the excess is allocated for the quantization information idwl2. A first normalization section and a first quantization section normalizes and quantizes a frequency spectrum mdspec1 in the first stage.
    Type: Grant
    Filed: August 25, 2015
    Date of Patent: May 2, 2017
    Assignee: SONY CORPORATION
    Inventors: Yuuki Matsumura, Shiro Suzuki, Keisuke Toyama, Mitsuyuki Hatanaka, Yuhki Mitsufuji