Vector Quantization Patents (Class 704/222)
  • Patent number: 11017786
    Abstract: Vector Quantizer and method therein for vector quantization, e.g. in a transform audio codec. The method comprises comparing an input target vector with four centroids C0, C1, C0,flip and C1,flip, wherein centroid C0,flip is a flipped version of centroid C0 and centroid C1,flip is a flipped version of centroid C1, each centroid representing a respective class of codevectors. A starting point for a search related to the input target vector in the codebook is determined, based on the comparison. A search is performed in the codebook, starting at the determined starting point, and a codevector is identified to represent the input target vector. A number of input target vectors per block or time segment is variable. A search space is dynamically adjusted to the number of input target vectors. The codevectors are sorted according to a distortion measure reflecting the distance between each codevector and the centroids C0 and C1.
    Type: Grant
    Filed: August 23, 2019
    Date of Patent: May 25, 2021
    Assignee: Telefonaktiebolaget LM Ericsson (publ)
    Inventors: Volodya Grancharov, Tomas Jansson Toftgård
  • Patent number: 10958318
    Abstract: The present disclosure relates to a 5G or pre-5G communication system for supporting a higher data transmission rate than in a 4G communication system such as LTE. The present disclosure relates to uplink transmission in a wireless communication system, and an operating method of a terminal includes mapping codes that are included in at least one codebook onto data symbols. and transmitting the data symbols spread by using the at least one codebook, and the data symbols are used for a base station to detect at least one active terminal including the terminal.
    Type: Grant
    Filed: August 3, 2018
    Date of Patent: March 23, 2021
    Assignees: Samsung Electronics Co., Ltd., Seoul National University R&DB Foundation
    Inventors: Sungnam Hong, Chanhong Kim, Sunho Park, Byonghyo Shim, Yeohun Yun, Seunghwan Lee, Guyoung Lim, Jongbu Lim, Hyoungju Ji, Taeyoung Kim
  • Patent number: 10929767
    Abstract: Embodiments of the present invention may provide the capability to detect complex events while providing improved detection and performance. In an embodiment of the present invention, a method for detecting an event may comprise receiving data representing measurement or detection of physical parameters, conditions, or actions, quantizing the received data and selecting a number of samples from the quantized data, generating a hidden Markov model representing events to be detected using initial model values based on ideal conditions, wherein a desired output is defined as a sequence of states, and wherein a number of states of the hidden Markov model is less than or equal to the number of samples of the quantized data, adjusting the quantized data and the initial model values to improve accuracy of the model, determining a state sequence of the hidden Markov model, and outputting an indication of a detected event.
    Type: Grant
    Filed: May 25, 2016
    Date of Patent: February 23, 2021
    Assignee: International Business Machines Corporation
    Inventors: Asaf Adi, Lior Limonad, Nir Mashkif, Segev E Wasserkrug, Alexander Zadorojniy, Sergey Zeltyn
  • Patent number: 10885894
    Abstract: Disclosed are a system and a method for singing expression transplantation. A singing expression transplantation method performed by a singing expression transplantation system according to an embodiment may comprise the steps of: synchronizing each of a first sound source and a second sound source, which include different pieces of voice information with regard to an identical song; modifying the pitch of the first sound source on the basis of pitch information extracted from each of the first sound source and the second sound source, which have been synchronized; and extracting volume information from each of the first sound source and the second sound source and adjusting the magnitude of the volume regarding the first sound source, the pitch of which has been modified, according to each piece of extracted volume information.
    Type: Grant
    Filed: December 15, 2017
    Date of Patent: January 5, 2021
    Assignee: Korea Advanced Institute of Science and Technology
    Inventors: Juhan Nam, Sangeon Yong
  • Patent number: 10878831
    Abstract: An apparatus includes a speech processing engine configured to receive data corresponding to speech and to determine whether a first characteristic associated with the speech differs from a reference characteristic by at least a threshold amount. The apparatus further includes a selection circuit responsive to the speech processing engine. The selection circuit is configured to select a particular speech codebook from among a plurality of speech codebooks based on the first characteristic differing from the reference characteristic by at least the threshold amount. The particular speech codebook is associated with the first characteristic.
    Type: Grant
    Filed: January 12, 2017
    Date of Patent: December 29, 2020
    Assignee: QUALCOMM Incorporated
    Inventors: Yinyi Guo, Erik Visser
  • Patent number: 10880275
    Abstract: Secure analytics using homomorphic and injective format-preserving encryption are disclosed herein. An example method includes encoding an analytic parameter set using a homomorphic encryption scheme as a set of homomorphic analytic vectors; transmitting the set of homomorphic analytic vectors to a server system; and receiving a homomorphic encrypted result from the server system, the server system having utilized the homomorphic encryption scheme and a first injective, format-preserving encryption scheme to evaluate the set of homomorphic analytic vectors over a datasource.
    Type: Grant
    Filed: January 19, 2018
    Date of Patent: December 29, 2020
    Assignee: Enveil, Inc.
    Inventor: Ellison Anne Williams
  • Patent number: 10811007
    Abstract: A computer-implemented method, according to one embodiment, includes: receiving a complex audio signal which includes an intended audio signal and at least one interfering audio signal. The complex audio signal is converted into text which represents a plurality of words included in the complex audio signal, and at least some of the text is identified as representing words which correspond to the at least one interfering audio signal. The identified text is discarded, and a remaining portion of the text is evaluated to determine whether the remaining portion of the text represents words which convey the voice-based command at an accuracy that is in a predetermined range. Furthermore, the remaining portion of the text is output in response to determining that the remaining portion of the text represents words which convey the voice-based command at an accuracy that is in the predetermined range.
    Type: Grant
    Filed: June 8, 2018
    Date of Patent: October 20, 2020
    Assignee: International Business Machines Corporation
    Inventors: Su Liu, Eric J. Rozner, Inseok Hwang, Chungkuk Yoo
  • Patent number: 10755731
    Abstract: A method for utterance section detection includes: executing pitch gain calculation processing that includes calculating a pitch gain indicating an intensity of periodicity of an audio signal expressing a voice of a speaker for each of frames that are obtained by dividing the audio signal and that each have a predetermined length; and executing utterance section detection processing that includes determining that an utterance section on the audio signal starts when the pitch gain becomes greater than or equal to a first threshold value after a non-utterance section on the audio signal lasts, wherein the utterance section detection processing further includes determining that the utterance section ends when the pitch gain becomes less than a second threshold value lower than the first threshold value after the utterance section lasts.
    Type: Grant
    Filed: July 7, 2017
    Date of Patent: August 25, 2020
    Assignee: FUJITSU LIMITED
    Inventors: Masanao Suzuki, Chisato Shioda, Nobuyuki Washio
  • Patent number: 10720165
    Abstract: A method of authenticating a user based on voice recognition of a keyword includes generating, at a processor, clean speech statistics. The clean speech statistics are generated from an audio recording of the keyword spoken by the user during an enrollment phase. The method further includes separating speech data and noise data from noisy input speech using the clean speech statistics during an authentication phase. The method also includes authenticating the user by comparing the speech data to the clean speech statistics or by comparing the noisy input speech to noisy speech statistics. The noisy speech statistics are based at least in part on the noise data.
    Type: Grant
    Filed: January 23, 2017
    Date of Patent: July 21, 2020
    Assignee: QUALCOMM Incorporated
    Inventors: Yinyi Guo, Erik Visser
  • Patent number: 10672388
    Abstract: A speech recognition system includes an input device to receive voice sounds, one or more processors, and one or more storage devices storing parameters and program modules including instructions which cause the one or more processors to perform operations.
    Type: Grant
    Filed: December 15, 2017
    Date of Patent: June 2, 2020
    Assignee: Mitsubishi Electric Research Laboratories, Inc.
    Inventors: Takaaki Hori, Shinji Watanabe, John Hershey
  • Patent number: 10614827
    Abstract: A speech-enhancing noise filter is disclosed. The noise filter comprises a microphone for acquiring speech data from a user; a feature extraction module configured to extract a plurality of features characterizing the speech data; a neural network configured to receive the plurality of extracted features and to estimate a noise profile from the plurality of extracted features; a noise removal module configured to remove the noise profile from the noisy speech data; and a reconstruction module configured to generate a waveform from the plurality of frames after removal of the noise profile from each of those frames. The neural network is trained to isolate various types of noise from the user speech in the speech data and then subtract the noise from the speech data, thus leaving only the user speech free of noise.
    Type: Grant
    Filed: February 21, 2018
    Date of Patent: April 7, 2020
    Inventor: Mohammad Mehdi Korjani
  • Patent number: 10580416
    Abstract: A method comprising: receiving lattice vector quantised parameter data, the parameter data representing at least one audio signal; determining within the data at least one bit error; and controlling the decoding of the data to generate an audio signal based on the determining of the bit error.
    Type: Grant
    Filed: July 6, 2015
    Date of Patent: March 3, 2020
    Assignee: Nokia Technologies Oy
    Inventors: Adriana Vasilache, Anssi Sakari Rämö, Lasse Juhani Laaksonen
  • Patent number: 10468044
    Abstract: Vector Quantizer and method therein for vector quantization, e.g. in a transform audio codec. The method comprises comparing an input target vector with four centroids C0, C1, C0,flip and C1,flip, wherein centroid C0,flip is a flipped version of centroid C0 and centroid C1,flip is a flipped version of centroid C1, each centroid representing a respective class of codevectors. A starting point for a search related to the input target vector in the codebook is determined, based on the comparison. A search is performed in the codebook, starting at the determined starting point, and a codevector is identified to represent the input target vector. A number of input target vectors per block or time segment is variable. A search space is dynamically adjusted to the number of input target vectors. The codevectors are sorted according to a distortion measure reflecting the distance between each codevector and the centroids C0 and C1.
    Type: Grant
    Filed: November 7, 2017
    Date of Patent: November 5, 2019
    Assignee: Telefonaktiebolaget LM Ericsson (publ)
    Inventors: Volodya Grancharov, Tomas Jansson Toftgård
  • Patent number: 10460738
    Abstract: Disclosed is an apparatus for processing an input signal, having a perceptual weighter and a quantizer. The perceptual weighter has a model provider and a model applicator. The model provider provides a perceptual weighted model based on the input signal. The model applicator provides a perceptually weighted spectrum by applying the perceptual weighted model to a spectrum based on the input signal. The quantizer is configured to quantize the perceptually weighted spectrum and for providing a bitstream. The quantizer has a random matrix applicator and a sign function calculator. The random matrix applicator is configured for applying a random matrix to the perceptually weighted spectrum in order to provide a transformed spectrum. The sign function calculator is configured for calculating a sign function of components of the transformed spectrum in order to provide the bitstream. The invention further refers to an apparatus for processing an encoded signal and to corresponding methods.
    Type: Grant
    Filed: March 13, 2017
    Date of Patent: October 29, 2019
    Assignee: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
    Inventors: Tom Baeckstroem, Florin Ghido, Johannes Fischer
  • Patent number: 10446173
    Abstract: An apparatus for speech processing: calculates a pitch gain indicating a magnitude of periodicity of an audio signal for each frame, the audio signal representing speaker's voice to be divided into the frames each having a predetermined length; determines that a speech production interval has started, when the pitch gain becomes equal or greater than a first threshold after a non-speech production interval; sets a second threshold that is lower than the first threshold by a reduction amount corresponding to a value acquired by subtracting a second representative value of the pitch gain in an interval preceding the start of the speech production interval from a first representative value of the pitch gain in the speech production interval; and determines that the speech production interval has terminated, when the pitch gain becomes smaller than the second threshold after the speech production interval has started.
    Type: Grant
    Filed: September 7, 2018
    Date of Patent: October 15, 2019
    Assignee: FUJITSU LIMITED
    Inventors: Chisato Shioda, Nobuyuki Washio, Masanao Suzuki
  • Patent number: 10409860
    Abstract: Systems, controllers and methods for contextual-based searching are provided. A system includes one or more devices configured to collect at least one audio stream and a contextual search system. The contextual search system includes a data analyzer and a search engine. The data analyzer is configured to receive the at least one audio stream from among the one or more devices and to determine contextual information from the received at least one audio stream. The search engine is configured to perform a search of at least one search term using the contextual information, to produce a search result.
    Type: Grant
    Filed: March 28, 2012
    Date of Patent: September 10, 2019
    Assignee: Staton Techiya, LLC
    Inventors: Steven W. Goldstein, Jordan Cohen
  • Patent number: 10402655
    Abstract: A system and method are provided for analyzing a video. The method comprises: sampling the video to generate a plurality of spatio-temporal video volumes; clustering similar ones of the plurality of spatio-temporal video volumes to generate a low-level codebook of video volumes; analyzing the low-level codebook of video volumes to generate a plurality of ensembles of volumes surrounding pixels in the video; and clustering the plurality of ensembles of volumes by determining similarities between the ensembles of volumes, to generate at least one high-level codebook. Multiple high-level codebooks can be generated by repeating steps of the method. The method can further include performing visual event retrieval by using the at least one high-level codebook to make an inference from the video, for example comparing the video to a dataset and retrieving at least one similar video, activity and event labeling, and performing abnormal and normal event detection.
    Type: Grant
    Filed: December 22, 2016
    Date of Patent: September 3, 2019
    Assignee: Sportlogiq Inc.
    Inventors: Mehrsan Javan Roshtkhari, Martin Levine
  • Patent number: 10403298
    Abstract: An information encoder for encoding an information signal includes: a converter for converting the linear prediction coefficients of the predictive polynomial A(z) to frequency values f1 . . . fn of a spectral frequency representation of the predictive polynomial A(z), wherein the converter is configured to determine the frequency values f1 . . . fn by analyzing a pair of polynomials P(z) and Q(z) being defined as P(z)=A(z)+z?m?lA(z?1) and Q(z)=A(z)?z?m?lA(z?1), wherein m is an order of the predictive polynomial A(z) and l is greater or equal to zero, wherein the converter is configured to obtain the frequency values by establishing a strictly real spectrum derived from P(z) and a strictly imaginary spectrum from Q(z) and by identifying zeros of the strictly real spectrum derived from P(z) and the strictly imaginary spectrum derived from Q(z).
    Type: Grant
    Filed: September 7, 2016
    Date of Patent: September 3, 2019
    Assignee: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
    Inventors: Tom Baeckstroem, Christian Fischer Pedersen, Johannes Fischer, Matthias Huettenberger, Alfonso Pino
  • Patent number: 10320453
    Abstract: A method of transmission over multiple wireless channels in a multiple antenna system includes storing channel modulation matrices at a transmitter; receiving quantized channel state information at the transmitter from plural receivers; selecting a transmission modulation matrix using the quantized channel state information from the stored channel modulation matrices; and transmitting over the multiple channels to the plural receivers using the selected transmission modulation matrix.
    Type: Grant
    Filed: February 23, 2015
    Date of Patent: June 11, 2019
    Assignee: WI-LAN INC.
    Inventors: Bartosz Mielczarek, Witold A. Krzymien
  • Patent number: 10304474
    Abstract: A method of enhancing speech quality includes: generating a high-frequency signal by using a low-frequency signal in a time domain; combining the low-frequency signal with the high-frequency signal; transforming the combined signal into a spectrum in a frequency domain; determining a class of a decoded speech signal; predicting an envelope from a low-frequency spectrum obtained in the transforming; and generating a final high-frequency spectrum by applying the predicted envelope to a high-frequency spectrum obtained in the transforming.
    Type: Grant
    Filed: August 17, 2015
    Date of Patent: May 28, 2019
    Assignee: SAMSUNG ELECTRONICS CO., LTD.
    Inventors: Ki-hyun Choo, Anton Viktorovich Porov, Konstantin Sergeevich Osipov, Eun-mi Oh, Woo-jung Park
  • Patent number: 10181327
    Abstract: A speech encoder that analyzes and classifies each frame of speech as being periodic-like speech or non-periodic like speech where the speech encoder performs a different gain quantization process depending if the speech is periodic or not. If the speech is periodic, the improved speech encoder obtains the pitch gains from the unquantized weighted speech signal and performs a pre-vector quantization of the adaptive codebook gain GP for each subframe of the frame before subframe processing begins and a closed-loop delayed decision vector quantization of the fixed codebook gain GC. If the frame of speech is non-periodic, the speech encoder may use any known method of gain quantization.
    Type: Grant
    Filed: March 6, 2009
    Date of Patent: January 15, 2019
    Assignee: Nytell Software LLC
    Inventors: Yang Gao, Adil Benyassine
  • Patent number: 10171202
    Abstract: A method of wirelessly communicating a packet can include generating, at a wireless device, a packet including a plurality of symbols. The method further includes segmenting an input bit vector into a plurality of symbol vectors according to one of a sequential or distributed segmentation procedure. The method further includes splitting each of the plurality of symbol vectors into two or more split vectors according to one of a sequential or round-robin split procedure. The method further includes mapping each of the split vectors into the plurality of symbols according to one of a block-level repetition or a symbol-level repetition. The method further includes transmitting the packet.
    Type: Grant
    Filed: November 6, 2015
    Date of Patent: January 1, 2019
    Assignee: QUALCOMM Incorporated
    Inventors: Lin Yang, Dung Ngoc Doan, Bin Tian
  • Patent number: 10170126
    Abstract: A method is provided for processing attenuation of pre-echo in a digital audio signal decoded by transform decoding. The method includes the following acts: decomposition of the decoded signal into at least two sub-signals according to a pre-determined decomposition criterion; calculation of attenuation factors per sub-signal and per sample of a previously determined pre-echo zone; attenuation of pre-echo in the pre-echo zone of each of the sub-signals by applying attenuation factors to the sub-signals; and production of the attenuated signal by addition of the attenuated sub-signals. Also provided are a processing device implementing the acts of the described method, and a decoder including such a device.
    Type: Grant
    Filed: December 20, 2013
    Date of Patent: January 1, 2019
    Assignee: ORANGE
    Inventors: Balazs Kovesi, Stephane Ragot
  • Patent number: 9966082
    Abstract: A spectrum filler for filling non-coded residual sub-vectors of a transform coded audio signal includes a sub-vector compressor configured to compress actually coded residual sub-vectors. A sub-vector rejecter is configured to reject compressed residual sub-vectors that do not fulfill a predetermined sparseness criterion. A sub-vector collector is configured to concatenate the remaining compressed residual sub-vectors to form a first virtual codebook. A coefficient combiner is configured to combine pairs of coefficients of the first virtual codebook to form a second virtual codebook. A sub-vector filler is configured to fill non-coded residual sub-vectors below a predetermined frequency with coefficients from the first virtual codebook, and to fill non-coded residual sub-vectors above the predetermined frequency with coefficients from the second virtual codebook.
    Type: Grant
    Filed: July 14, 2016
    Date of Patent: May 8, 2018
    Assignee: TELEFONAKTIEBOLAGET LM ERICSSON (PUBL)
    Inventors: Volodya Grancharov, Sebastian Näslund, Sigurdur Sverrisson
  • Patent number: 9946511
    Abstract: Provided is a method for user training of an information dialog system. The method may include activating a user input subsystem, receiving a training request entered by the user, converting the training request into text by the user input subsystem, sending the text of the training request obtained as a result of the conversion to a dialog module, processing the text of the training request by the dialog module, forming a response to the training request by the dialog module, and sending the response to the training request to the user. The response to the training request may be formed in a form of one or more of the following: a voice cue, a text, and an action performed by the information dialog system.
    Type: Grant
    Filed: May 26, 2015
    Date of Patent: April 17, 2018
    Assignee: GOOGLE LLC
    Inventors: Ilya Genadevich Gelfenbeyn, Olga Aleksandrovna Gelfenbeyn, Artem Goncharuk, Ilya Andreevich Platonov, Pavel Aleksandrovich Sirotin
  • Patent number: 9928838
    Abstract: In an embodiment, a system on a chip (SOC) may include one or more central processing units (CPUs), a memory controller, and a circuit configured to remain powered on when the rest of the SOC is powered down. The circuit may be configured to receive audio samples and match those audio samples against a predetermined pattern. The circuit may operate according to a first clock during the time that the rest of the SOC is powered down. In response to detecting the predetermined pattern in the samples, the circuit may cause the memory controller and processors to power up. During the power up process, a second clock having one or more better characteristics than the first clock may become available. The circuit may switch to the second clock while preserving the samples, or losing at most one sample, or no more than a threshold number of samples.
    Type: Grant
    Filed: April 7, 2017
    Date of Patent: March 27, 2018
    Assignee: Apple Inc.
    Inventors: Manu Gulati, Gilbert H. Herbeck, Alexei E. Kosut, Girault W. Jones, Timothy J. Millet
  • Patent number: 9892742
    Abstract: An apparatus comprising: a vector generator configured to generate at least one vector of parameters defining at least one audio signal; a lattice vector quantizer configured to sort the at least one vector of parameters according to an ordering of at least one vector absolute tuples to generate an associated at least one ordered vector of parameters; the lattice vector quantizer configured to select from a list of leader classes at least one potential code vector; the lattice vector quantizer configured to determine a distance between the at least one potential code vector and the at least one ordered vector of parameters; the lattice vector quantizer configured to determine at least one leader class associated with a potential code vector which generates the smallest associated distance; the lattice vector quantizer configured to transpose the at least one leader class to generate an output lattice quantized codevector.
    Type: Grant
    Filed: December 17, 2013
    Date of Patent: February 13, 2018
    Assignee: Nokia Technologies Oy
    Inventors: Adriana Vasilache, Anssi Sakari Rämö, Lasse Juhani Laaksonen
  • Patent number: 9881620
    Abstract: Provided are, among other things, systems, methods and techniques for compressing an audio signal. According to one representative embodiment, an audio signal that includes quantization indexes, identification of segments of such quantization indexes, and indexes of entropy codebooks that have been assigned to such segments is obtained, with a single entropy codebook index having been assigned to each such segment. Potential merging operations in which adjacent ones of the segments potentially would be merged with each are identified, and bit penalties for the potential merging operations are estimated. At least one of the potential merging operations is performed based on the estimated bit penalties, thereby obtaining a smaller updated set of segments of quantization indexes and corresponding assigned codebooks. The quantization indexes in each of the segments in the smaller updated set are then entropy encoded by using the corresponding assigned entropy codebooks, thereby compressing the audio signal.
    Type: Grant
    Filed: December 4, 2016
    Date of Patent: January 30, 2018
    Assignee: Digital Rise Technology Co., Ltd.
    Inventor: Yuli You
  • Patent number: 9875748
    Abstract: A noise attenuation apparatus receives an audio signal comprising a desired and a noise signal component. Two codebooks (109, 111) comprise respectively desired signal candidates representing a possible desired signal component and noise signal contribution candidates representing possible noise contributions. A segmenter (103) segments the audio signal into time segments and for each time segment a noise attenuator (105) generates estimated signal candidates by for each of the desired signal candidates generating an estimated signal candidate as a combination of a scaled version of the desired signal candidate and a weighted combination of the noise signal contribution candidates. The noise attenuator (105) minimizes a cost function indicative of a difference between the estimated signal candidate and the audio signal in the time segment. A signal candidate is then determined for the time segment from the estimated signal candidates and the audio signal is noise compensated based on this signal candidate.
    Type: Grant
    Filed: October 22, 2012
    Date of Patent: January 23, 2018
    Assignee: KONINKLIJKE PHILIPS N.V.
    Inventor: Sriram Srinivasan
  • Patent number: 9830920
    Abstract: A method, device, and apparatus provide the ability to predict a portion of a polyphonic audio signal for compression and networking applications. The solution involves a framework of a cascade of long term prediction filters, which by design is tailored to account for all periodic components present in a polyphonic signal. This framework is complemented with a design method to optimize the system parameters. Specialization may include specific techniques for coding and networking scenarios, where the potential of each enhanced prediction is realized to considerably improve the overall system performance for that application. One specific technique provides enhanced inter-frame prediction for the compression of polyphonic audio signals, particularly at low delay. Another specific technique provides improved frame loss concealment capabilities to combat packet loss in audio communications.
    Type: Grant
    Filed: June 29, 2016
    Date of Patent: November 28, 2017
    Assignee: The Regents of the University of California
    Inventors: Kenneth Rose, Tejaswi Nanjundaswamy
  • Patent number: 9792922
    Abstract: An encoder and a method therein for Pyramid Vector Quantizer, PVQ, shape search, the PVQ taking a target vector x as input and deriving a vector y by iteratively adding unit pulses in an inner dimension search loop. The method comprises, before entering a next inner dimension search loop for unit pulse addition, determining, based on the maximum pulse amplitude, maxampy, of a current vector y, whether more than a current bit word length is needed to represent enloopy, in a lossless manner in the upcoming inner dimension loop. The variable enloopy is related to an accumulated energy of the vector y. The performing of this method enables the encoder to keep the complexity of the search at a reasonable level.
    Type: Grant
    Filed: June 25, 2015
    Date of Patent: October 17, 2017
    Assignee: TELEFONAKTIEBOLAGET LM ERICSSON (PUBL)
    Inventor: Jonas Svedberg
  • Patent number: 9747908
    Abstract: An audio signal decoding apparatus is provided that includes a receiver that receives an encoded information, a memory, and a processor that demultiplexes low-band encoding parameters, index information, and scale factor information from the encoded information. The processor also decodes the low-band encoding parameters to obtain a synthesized low frequency spectrum, replicates a high frequency subband spectrum based on the index information using the synthesized low frequency spectrum, and adjusts an amplitude of the replicated high frequency subband spectrum using the scale factor information. The processor further estimates a frequency of a harmonic component in the synthesized low frequency spectrum, adjusts a frequency of a harmonic component in the high frequency subband spectrum using the estimated harmonic frequency spectrum, and generates an output signal using the synthesized low frequency spectrum and the high frequency subband spectrum.
    Type: Grant
    Filed: October 5, 2016
    Date of Patent: August 29, 2017
    Assignee: PANASONIC INTELLECTUAL PROPERTY CORPORATION OF AMERICA
    Inventors: Srikanth Nagisetty, Zongxian Liu
  • Patent number: 9711150
    Abstract: An audio encoding apparatus to encode an audio signal using lossless coding or lossy coding and an audio decoding apparatus to decode an encoded audio signal are disclosed. An audio encoding apparatus according to an exemplary embodiment may include an input signal type determination unit to determine a type of an input signal based on characteristics of the input signal, a residual signal generation unit to generate a residual signal based on an output signal from the input signal type determination unit, and a coding unit to perform lossless coding or lossy coding using the residual signal.
    Type: Grant
    Filed: August 22, 2013
    Date of Patent: July 18, 2017
    Assignees: Electronics and Telecommunications Research Institute, The Korea Development Bank
    Inventors: Seung Kwon Beack, Tae Jin Lee, Kyeong Ok Kang, Keun Woo Choi, Jong Mo Sung
  • Patent number: 9679555
    Abstract: A method for measuring speech signal quality by an electronic device is described. The method includes obtaining a modified single-channel speech signal. The method also includes estimating multiple objective distortions based on the modified single-channel speech signal. The multiple objective distortions include at least one foreground distortion and at least one background distortion. The method further includes estimating a foreground quality and a background quality based on the multiple objective distortions. The method additionally includes estimating an overall quality based on the foreground quality and the background quality.
    Type: Grant
    Filed: June 24, 2014
    Date of Patent: June 13, 2017
    Assignee: QUALCOMM Incorporated
    Inventors: Dipanjan Sen, Wenliang Lu
  • Patent number: 9652625
    Abstract: Disclosed are systems and methods for counteracting unauthorized access to microphone data. An example method include storing, in a data buffer, audio data received from an audio endpoint device, installing, a software driver associated with the audio session, where the software driver prevents access to the audio data by unauthorized software applications, and receiving process identifier data from a software application requesting to access the audio data stored in the data buffer. Furthermore, the method includes determining whether the application requesting access to the audio data is an unauthorized software application and controlling the software driver to prevent access to the audio data by the determined unauthorized software application.
    Type: Grant
    Filed: July 5, 2016
    Date of Patent: May 16, 2017
    Assignee: AO Kaspersky Lab
    Inventors: Vyacheslav I. Levchenko, Alexander V. Kalinin
  • Patent number: 9653079
    Abstract: In an embodiment, a system on a chip (SOC) may include one or more central processing units (CPUs), a memory controller, and a circuit configured to remain powered on when the rest of the SOC is powered down. The circuit may be configured to receive audio samples and match those audio samples against a predetermined pattern. The circuit may operate according to a first clock during the time that the rest of the SOC is powered down. In response to detecting the predetermined pattern in the samples, the circuit may cause the memory controller and processors to power up. During the power up process, a second clock having one or more better characteristics than the first clock may become available. The circuit may switch to the second clock while preserving the samples, or losing at most one sample, or no more than a threshold number of samples.
    Type: Grant
    Filed: February 12, 2015
    Date of Patent: May 16, 2017
    Assignee: Apple Inc.
    Inventors: Manu Gulati, Gilbert H. Herbeck, Alexei E. Kosut, Girault W. Jones, Timothy J. Millet
  • Patent number: 9619551
    Abstract: A computer-implemented system and method for generating document groupings is provided. A lexicon of terms extracted from a set of documents is generated. The lexicon includes a frequency of each extracted term within each document in the set. Concepts each having two or more of the extracted terms are generated. A subset of the documents in the set is selected based on the term frequencies. The subset of documents is grouped into clusters based on the concepts. A similarity of each document cluster is calculated with one or more documents based on a distance by summing the frequency of each term in that document and a weight of the cluster for each of the terms. The weights are updated until a rate of change for each cluster becomes constant.
    Type: Grant
    Filed: November 23, 2015
    Date of Patent: April 11, 2017
    Assignee: FTI Technology LLC
    Inventors: Dan Gallivan, Kenji Kawai
  • Patent number: 9584833
    Abstract: Improved methods for coding an ensemble of pulse vectors utilize statistical models (i.e., probability models) for the ensemble of pulse vectors, to more efficiently code each pulse vector of the ensemble. At least one pulse parameter describing the non-zero pulses of a given pulse vector is coded using the statistical models and the number of non-zero pulse positions for the given pulse vector. In some embodiments, the number of non-zero pulse positions are coded using range coding. The total number of unit magnitude pulses may be coded using conditional (state driven) bitwise arithmetic coding. The non-zero pulse position locations may be coded using adaptive arithmetic coding. The non-zero pulse position magnitudes may be coded using probability-based combinatorial coding, and the corresponding sign information may be coded using bitwise arithmetic coding. Such methods are well suited to coding non-independent-identically-distributed signals, such as coding video information.
    Type: Grant
    Filed: May 3, 2016
    Date of Patent: February 28, 2017
    Assignee: Google Technology Holdings LLC
    Inventors: James P. Ashley, Udar Mittal
  • Patent number: 9558754
    Abstract: In one embodiment, an audio decoder for decoding an encoded audio bitstream is disclosed. The audio decoder is capable of being operated in at least three different decoding modes. The audio decoder includes a demultiplexer for obtaining audio data and control information from the encoded audio bitstream. The audio decoder also includes a first audio decoder configured to operate in a first decoding mode using a first decoding technique and a second audio decoder configured to operate in a second decoding mode using a second decoding technique. The audio decoder also includes a pitch predictor integrated into the second audio decoder. The pitch predictor includes a long-term prediction filter and a short-term prediction filter. The audio decoder further includes a selector for selecting one of the at least three different decoding modes based on at least some of the control information.
    Type: Grant
    Filed: April 12, 2016
    Date of Patent: January 31, 2017
    Assignee: Dolby International AB
    Inventors: Barbara Resch, Kristofer Kjörling, Lars Villemoes
  • Patent number: 9489959
    Abstract: The purpose of the present invention is to more efficiently extend, using a low bit rate, the bandwidth of input signals having a harmonics structure, in order to obtain better audio quality. The present invention is installed in a device that extends bandwidth for audio signal encoding and decoding. This novel bandwidth extension encoding identifies a low-frequency spectrum component having the highest correlation to a high-frequency bandwidth signal among input signals, duplicates a high-frequency spectrum by energy adjustment of said component, and maintains the harmonic relationship between the low-frequency spectrum and the duplicated high-frequency spectrum by adjusting the spectral peak position of the duplicated high-frequency spectrum, on the basis of a harmonic frequency estimated from a composite low-frequency spectrum.
    Type: Grant
    Filed: June 10, 2014
    Date of Patent: November 8, 2016
    Assignee: PANASONIC INTELLECTUAL PROPERTY CORPORATION OF AMERICA
    Inventors: Srikanth Nagisetty, Zongxian Liu
  • Patent number: 9418342
    Abstract: A method, computer-readable storage device and apparatus for determining a mode of motion are disclosed. For example, a method receives training data comprising gait information associated with a plurality of different modes of motion. The method performs principal component analysis on the training data to extract principal components from the training data and generates a hidden markov model for each of a plurality of different modes of motion based upon the training data. The method receives testing data comprising gait information, transforms the testing data based upon the principal components and calculates a likelihood of the testing data based upon each hidden markov model for each of the plurality of different modes of motion. The method determines the mode of motion of the testing data, where the mode of motion is one of the plurality of different modes of motion for which a highest likelihood is calculated.
    Type: Grant
    Filed: December 6, 2013
    Date of Patent: August 16, 2016
    Assignees: AT&T Intellectual Property I, L.P., President and Fellows of Harvard College
    Inventors: Saeed S. Ghassemzadeh, Lusheng Ji, Robert Raymond Miller, II, Manish Gupta, Vahid Tarokh
  • Patent number: 9386386
    Abstract: Systems and methods for enhancing the audio experience on a consumer electronic device are disclosed. More particularly systems and methods for optimizing the audio performance of individual consumer electronic devices as part of a manufacturing process and/or retail experience are disclosed. A system for enhancing the audio performance of a consumer electronic device including a parametrically configurable processing block is disclosed.
    Type: Grant
    Filed: January 9, 2013
    Date of Patent: July 5, 2016
    Assignee: Actiwave AB
    Inventors: Pär Gunnars Risberg, Richard Kjerstadius, Landy Toth
  • Patent number: 9361894
    Abstract: A low bit rate digital audio coding system includes an encoder which assigns codebooks to groups of quantization indexes based on their local properties resulting in codebook application ranges that are independent of block quantization boundaries. The invention also incorporates a resolution filter bank, or a tri-mode resolution filter bank, which is selectively switchable between high and low frequency resolution modes or high, low and intermediate modes such as when detecting transient in a frame. The result is a multichannel audio signal having a significantly lower bit rate for efficient transmission or storage. The decoder is essentially an inverse of the structure and methods of the encoder, and results in a reproduced audio signal that cannot be audibly distinguished from the original signal.
    Type: Grant
    Filed: May 15, 2013
    Date of Patent: June 7, 2016
    Assignee: Digital Rise Technology Co., Ltd.
    Inventor: Yuli You
  • Patent number: 9336773
    Abstract: Disclosed herein are systems, methods, and computer-readable storage media for selecting a speech recognition model in a standardized speech recognition infrastructure. The system receives speech from a user, and if a user-specific supervised speech model associated with the user is available, retrieves the supervised speech model. If the user-specific supervised speech model is unavailable and if an unsupervised speech model is available, the system retrieves the unsupervised speech model. If the user-specific supervised speech model and the unsupervised speech model are unavailable, the system retrieves a generic speech model associated with the user. Next the system recognizes the received speech from the user with the retrieved model. In one embodiment, the system trains a speech recognition model in a standardized speech recognition infrastructure. In another embodiment, the system handshakes with a remote application in a standardized speech recognition infrastructure.
    Type: Grant
    Filed: May 1, 2015
    Date of Patent: May 10, 2016
    Assignee: INTERACTIONS LLC
    Inventors: Andrej Ljolje, Bernard S. Renger, Steven Neil Tischer
  • Patent number: 9299347
    Abstract: Methods, systems, and apparatus are described that receive audio data for an utterance. Association data is accessed that indicates associations between data corresponding to uncorrupted audio segments, and data corresponding to corrupted versions of the uncorrupted audio segments, where the associations are determined before receiving the audio data for the utterance. Using the association data and the received audio data for the utterance, data corresponding to at least one uncorrupted audio segment is selected. A transcription of the utterance is determined based on the selected data corresponding to the at least one uncorrupted audio segment.
    Type: Grant
    Filed: April 14, 2015
    Date of Patent: March 29, 2016
    Assignee: Google Inc.
    Inventors: Olivier Siohan, Pedro J. Moreno Mengibar
  • Patent number: 9294113
    Abstract: Described herein is a sampling system and related sampling scheme. The system and sampling scheme is based upon a framework for adaptive non-uniform sampling schemes. In the system and schemes described herein, time intervals between samples can be computed by using a function of previously taken samples. Therefore, keeping sampling times (time-stamps), except initialization times, is not necessary. One aim of this sampling framework is to provide a balance between reconstruction distortion and average sampling rate. The function by which sampling time intervals can be computed is called the sampling function. The sampling scheme described herein can be applied appropriately on different signal models such as deterministic or stochastic, and continuous or discrete signals. For each different signal model, sampling functions can be derived.
    Type: Grant
    Filed: July 5, 2012
    Date of Patent: March 22, 2016
    Assignee: Massachusetts Institute of Technology
    Inventors: Soheil Feizi-Khankandi, Vivek K. Goyal, Muriel Médard
  • Patent number: 9268762
    Abstract: According to various embodiments of the disclosure techniques for generating outgoing messages are disclosed. The technique includes receiving a request to generate an outgoing message for a recipient and retrieving one or more recipient preferences of the recipient from a recipient preferences database. The one or more recipient preferences relate to customization of messages that are to be delivered to the recipient. The technique further includes retrieving a message template from a plurality of message templates stored in a message template database based on the request and the one or more recipient preferences. The technique also includes generating the outgoing message based on the retrieved message template and the one or more recipient preferences, and providing the outgoing message to the recipient.
    Type: Grant
    Filed: January 16, 2012
    Date of Patent: February 23, 2016
    Assignee: Google Inc.
    Inventors: Kirill Buryak, Andrew Swerdlow, Luke Hiro Swartz, Cibu Chalissery Johny
  • Patent number: 9269365
    Abstract: There is provided a method of encoding an input speech signal. The method comprises identifying a fixed codebook vector from a fixed codebook; identifying an adaptive codebook vector from a adaptive codebook; calculating an adaptive codebook gain; reducing the adaptive codebook gain by an amount; optimally selecting a fixed codebook gain based on the adaptive codebook gain while both the fixed codebook vector and the adaptive codebook vector remain fixed; and converting the input speech signal into an encoded speech using the fixed codebook gain, the adaptive codebook gain, the fixed codebook vector and the adaptive codebook vector. The amount of reducing the adaptive codebook gain may be varied.
    Type: Grant
    Filed: July 11, 2008
    Date of Patent: February 23, 2016
    Assignee: Mindspeed Technologies, Inc.
    Inventors: Huan-Yu Su, Yang Gao
  • Patent number: 9202473
    Abstract: It is inter alia disclosed to determine, for each set of basis code vectors of a plurality of sets of basis code vectors, a potential basis code vector for encoding an input vector, wherein each set of basis code vectors is associated with at least one scale representative of a plurality of scale representatives, and to determine a code vector for encoding the input vector from a subset of code vectors, said subset of code vectors comprising, for each determined potential basis code vector and each scale representative associated with the set of basis code vectors of the respective potential basis code vector, a code vector based on the respective potential basis code vector scaled by the respective scale representative.
    Type: Grant
    Filed: July 1, 2011
    Date of Patent: December 1, 2015
    Assignee: Nokia Technologies Oy
    Inventors: Adriana Vasilache, Lasse Juhani Laaksonen, Anssi Sakari Rämö, Mikko Tapio Tammi
  • Patent number: 9165563
    Abstract: For respective sampling data of waveform data of sounds to be coded, a prediction residual value is calculated as sampling residual data, and an effective bit length is calculated from this residual waveform data. Then, for the effective bit length data, a maximum effective bit length among processing targets is generated as common effective actual data, and coded data in which this common effective actual data and information indicating the common effective bit length are arranged in a predetermined configuration format are generated. The information included in the coded data is analyzed and each of the plurality of the common effective bit information is extracted. Then, waveform data of the sounds are decoded by performing inverse linear prediction processing from an analysis result on the residual waveform data decompressed by performing bit extension which adds a portion other than the common effective bit length.
    Type: Grant
    Filed: December 26, 2012
    Date of Patent: October 20, 2015
    Assignee: CASIO COMPUTER CO., LTD.
    Inventor: Goro Sakata