Vector Quantization Patents (Class 704/222)
  • 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: 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: 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: 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: 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: 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: 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
  • Patent number: 9153238
    Abstract: The present invention relates to a method for processing an audio signal, comprising the following steps: performing a linear predictive analysis on the current frame of an audio signal so as to generate a first target vector, which is a target vector of a first stage, on the basis of a plurality of linear prediction transform coefficients; performing vector quantization on the first target vector so as to acquire a predetermined number of first temporary candidate code vectors of the first stage; calculating first temporary candidate errors, which are errors between the first temporary candidate code vectors and the first target vector; and determining a first number, which is the number of the first candidate code vectors, on the basis of the first temporary candidate errors, and acquiring first final candidate code vectors in the same amount as the first number.
    Type: Grant
    Filed: April 8, 2011
    Date of Patent: October 6, 2015
    Assignee: LG Electronics Inc.
    Inventors: Gyu Hyeok Jeong, Hye Jeong Jeon, Byung Suk Lee, Chang Heon Lee
  • Patent number: 9154849
    Abstract: Disclosed are a transmitter and a receiver for providing multi-layered multimedia services, and a method thereof. The transmitter for providing multimedia services includes a multiple description coding unit that performs multiple description coding (MDC) with respect to at least one source to thereby output a description sequence with respect to each of the at least one source, a description multiplexing unit that multiplexes the description sequence in units of descriptions to thereby output a single multiplexing description sequence, and a transmission code block processing unit that divides and modulates the single multiplexing description sequence to thereby generate a transmission block. Therefore, it is possible to provide high-quality multimedia services while ensuring graceful performance degradation and scalability.
    Type: Grant
    Filed: December 20, 2012
    Date of Patent: October 6, 2015
    Assignee: Electronics and Telecommunications Research Institute
    Inventor: Seong Rag Kim
  • Patent number: 9135919
    Abstract: A quantization device and quantization method are provided that reduce coding distortion with a small degree of calculation and achieve adequate coding performance thereby. A multistage vector quantization unit treats a number of candidates N that are designated prior to operation in the first-stage vector quantization unit, decrements the number of candidates by one beginning with the second-stage vector quantization unit and continuing with each stage thereafter. If the number of candidates is three or less, the multistage vector quantization unit assesses the quantization distortion at each stage, treating the number of candidates at the following stage as a predetermined value P if the quantization distortion is greater than a prescribed threshold, and treating the number of candidates at the following stage as a value Q that is less than the predetermined value P if the quantization distortion is less than or equal to the predetermined threshold.
    Type: Grant
    Filed: September 16, 2011
    Date of Patent: September 15, 2015
    Assignee: PANASONIC INTELLECTUAL PROPERTY CORPORATION OF AMERICA
    Inventor: Toshiyuki Morii
  • Patent number: 9129597
    Abstract: An audio signal decoder configured to provide a decoded audio signal representation on the basis of an encoded audio signal representation including a sampling frequency information, an encoded time warp information and an encoded spectrum representation includes a time warp calculator and a warp decoder. The time warp calculator is configured to adapt a mapping rule for mapping codewords of the encoded time warp information onto decoded time warp values describing the decoded time warp information in dependence on the sampling frequency information. The warp decoder is configured to provide the decoded audio signal representation on the basis of the encoded spectrum representation and in dependence on the decoded time warp information.
    Type: Grant
    Filed: September 6, 2012
    Date of Patent: September 8, 2015
    Assignees: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e. V., Dolby International AB
    Inventors: Stefan Bayer, Tom Baeckstroem, Ralf Geiger, Bernd Edler, Sascha Disch, Lars Villemoes
  • Patent number: 9100053
    Abstract: In a method for reconstructing a source signal, which is encoded by a set of at least two descriptions, the method comprises: receiving a subset of the set of descriptions; reconstructing a reconstructed signal at an operating bitrate of a set of operating bitrates upon the basis of the subset of descriptions, the reconstructed signal having a second probability density, wherein the second probability density comprises a first statistical moment and a second statistical moment; and manipulating the reconstructed signal, wherein the reconstructed signal is manipulated such that, irrespective of the operating bitrate, a predetermined minimum similarity between the first statistical moment of the third probability density and the first statistical moment of the first probability density and between the second statistical moment of the third probability density and the second statistical moment of the first probability density is maintained.
    Type: Grant
    Filed: November 22, 2013
    Date of Patent: August 4, 2015
    Assignee: Huawei Technologies Co., Ltd.
    Inventors: Janusz Klejsa, Guoqiang Zhang, Minyue Li, Willem Bastiaan Kleijn
  • Patent number: 9099099
    Abstract: System and method embodiments are provided for very short pitch detection and coding for speech or audio signals. The system and method include detecting whether there is a very short pitch lag in a speech or audio signal that is shorter than a conventional minimum pitch limitation using a combination of time domain and frequency domain pitch detection techniques. The pitch detection techniques include using pitch correlations in time domain and detecting a lack of low frequency energy in the speech or audio signal in frequency domain. The detected very short pitch lag is coded using a pitch range from a predetermined minimum very short pitch limitation that is smaller than the conventional minimum pitch limitation.
    Type: Grant
    Filed: December 21, 2012
    Date of Patent: August 4, 2015
    Assignee: Huawei Technologies Co., Ltd.
    Inventors: Yang Gao, Fengyan Qi
  • Patent number: 9088323
    Abstract: A method of reporting a channel state to a base station supporting downlink multiple input multiple output in a wireless communication system includes measuring a downlink channel based on a downlink signal received from the base station, and reporting a codebook index for precoding of the downlink signal according to the measurement of the downlink channel. The codebook for precoding includes an improved codebook (hereinafter, referred to as an improved 4Tx codebook) having a dual structure composed of a codebook for wideband and a codebook for subband, for 4 transmit (4Tx) antenna ports, and the improved 4Tx codebook includes sub-codebooks obtained by sub-sampling the codebook for subband according to priority.
    Type: Grant
    Filed: January 8, 2014
    Date of Patent: July 21, 2015
    Assignee: LG Electronics Inc.
    Inventors: Hanjun Park, Youngtae Kim, Kijun Kim, Hyungtae Kim
  • Patent number: 9076433
    Abstract: An apparatus for generating a synthesis audio signal using a patching control signal has a first converter, a spectral domain patch generator, a high frequency reconstruction manipulator and a combiner. The first converter is configured for converting a time portion of an audio signal into a spectral representation. The spectral domain patch generator is configured for performing a plurality of different spectral domain patching algorithms, wherein each patching algorithm generates a modified spectral representation having spectral components in an upper frequency band derived from corresponding spectral components in a core frequency band of the audio signal.
    Type: Grant
    Filed: November 28, 2012
    Date of Patent: July 7, 2015
    Assignee: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
    Inventors: Frederik Nagel, Markus Multrus, Jeremie Lecomte, Stefan Bayer, Guillaume Fuchs, Johannes Hilpert, Julien Robilliard
  • Patent number: 9076442
    Abstract: According to the present invention, a linear prediction filter coefficient of a current frame is acquired from an input signal using linear prediction, a quantized spectrum candidate vector of the current frame, corresponding to the linear prediction filter coefficient of the current frame, is acquired on the basis of first best information, and the quantized spectrum candidate vector of the current frame and the quantized spectrum vector of the previous frame are interpolated. Accordingly, in contrast to conventional phased optimization techniques, optimum parameters which minimize quantization errors, can be obtained.
    Type: Grant
    Filed: December 10, 2010
    Date of Patent: July 7, 2015
    Assignees: LG Electronics Inc., Industry-Academic Cooperation Foundation, Yonsei University
    Inventors: Hyejeong Jeon, Daehwan Kim, Gyuhyeok Jeong, Minki Lee, Honggoo Kang, Byungsuk Lee, Lagyoung Kim
  • Patent number: 9053704
    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: July 14, 2014
    Date of Patent: June 9, 2015
    Assignee: Interactions LLC
    Inventors: Andrej Ljolje, Bernard S. Renger, Steven Neil Tischer
  • Patent number: 9031840
    Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for receiving (i) audio data that encodes a spoken natural language query, and (ii) environmental audio data, obtaining a transcription of the spoken natural language query, determining a particular content type associated with one or more keywords in the transcription, providing at least a portion of the environmental audio data to a content recognition engine, and identifying a content item that has been output by the content recognition engine, and that matches the particular content type.
    Type: Grant
    Filed: December 27, 2013
    Date of Patent: May 12, 2015
    Assignee: Google Inc.
    Inventors: Matthew Sharifi, Gheorghe Postelnicu
  • Patent number: 9026434
    Abstract: An audio coding terminal and method is provided. The terminal includes a coding mode setting unit to set an operation mode, from plural operation modes, for input audio coding by a codec configured to code the input audio based on the set operation mode such that when the set operation mode is a high frame erasure rate (FER) mode the codec codes a current frame of the input audio according to a select frame erasure concealment (FEC) mode of one or more FEC modes. Upon the setting of the operation mode to be the High FER mode the one FEC mode is selected, from the one or more FEC modes predetermined for the High FER mode, to control the codec by incorporating of redundancy within a coding of the input audio or as separate redundancy information separate from the coded input audio according to the selected one FEC mode.
    Type: Grant
    Filed: April 10, 2012
    Date of Patent: May 5, 2015
    Assignee: Samsung Electronic Co., Ltd.
    Inventors: Steven Craig Greer, Hosang Sung
  • Patent number: 9015040
    Abstract: An apparatus for encoding an audio signal having a stream of audio samples has: a windower for applying a prediction coding analysis window to the stream of audio samples to obtain windowed data for a prediction analysis and for applying a transform coding analysis window to the stream of audio samples to obtain windowed data for a transform analysis, wherein the transform coding analysis window is associated with audio samples within a current frame of audio samples and with audio samples of a predefined portion of a future frame of audio samples being a transform-coding look-ahead portion, wherein the prediction coding analysis window is associated with at least the portion of the audio samples of the current frame and with audio samples of a predefined portion of the future frame being a prediction coding look-ahead portion, wherein the transform coding look-ahead portion and the prediction coding look-ahead portion are identically to each other or are different from each other by less than 20%; and an enc
    Type: Grant
    Filed: August 14, 2013
    Date of Patent: April 21, 2015
    Assignee: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
    Inventors: Emmanuel Ravelli, Ralf Geiger, Markus Schnell, Guillaume Fuchs, Vesa Ruoppila, Tom Baeckstroem, Bernhard Grill, Christian Helmrich
  • Patent number: 9015041
    Abstract: An audio encoder has a window function controller, a windower, a time warper with a final quality check functionality, a time/frequency converter, a TNS stage or a quantizer encoder, the window function controller, the time warper, the TNS stage or an additional noise filling analyzer are controlled by signal analysis results obtained by a time warp analyzer or a signal classifier. Furthermore, a decoder applies a noise filling operation using a manipulated noise filling estimate depending on a harmonic or speech characteristic of the audio signal.
    Type: Grant
    Filed: January 11, 2011
    Date of Patent: April 21, 2015
    Assignee: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
    Inventors: Stefan Bayer, Sascha Disch, Ralf Geiger, Guillaume Fuchs, Max Neuendorf, Gerald Schuller, Bernd Edler
  • Patent number: 9009037
    Abstract: Disclosed is an encoding device that improves the quality of a decoded signal in a hierarchical coding (scalable coding) method, wherein a band to be quantized is selected for every level (layer). The encoding device (101) is equipped with a second layer encoding unit (205) that selects a first band to be quantized of a first input signal from among a plurality of sub-bands, and that generates second layer encoding information containing first band information of said band; a second layer decoding unit (206) that generates a first decoded signal using the second layer encoding information; an addition unit (207) that generates a second input signal using the first input signal and the first decoded signal; and a third layer encoding unit (208) that selects a second band to be quantized of the second input signal using the first decoded signal, and that generates third layer encoding information.
    Type: Grant
    Filed: October 13, 2010
    Date of Patent: April 14, 2015
    Assignee: Panasonic Intellectual Property Corporation of America
    Inventors: Tomofumi Yamanashi, Toshiyuki Morii
  • Patent number: 8990076
    Abstract: In automated speech recognition (ASR), multiple devices may be employed to perform the ASR in a distributed environment. To reduce bandwidth use in transmitting between devices ASR information is compressed prior to transmission. To counteract fidelity loss that may accompany such compression, two versions of an audio signal are processed by an acoustic front end (AFE), one version is unaltered and one is compressed and decompressed prior to AFE processing. The two versions are compared, and the comparison data is sent to a recipient for further ASR processing. The recipient uses the comparison data and a received version of the compressed audio signal to recreate the post-AFE processing results from the received audio signal. The result is improved ASR results and decreased bandwidth usage between distributed ASR devices.
    Type: Grant
    Filed: September 10, 2012
    Date of Patent: March 24, 2015
    Assignee: Amazon Technologies, Inc.
    Inventor: Nikko Strom
  • Patent number: 8977544
    Abstract: A quantizing method is provided that includes quantizing an input signal by selecting one of a first quantization scheme not using an inter-frame prediction and a second quantization scheme using the inter-frame prediction, in consideration of one or more of a prediction mode, a predictive error and a transmission channel state.
    Type: Grant
    Filed: April 23, 2012
    Date of Patent: March 10, 2015
    Assignee: Samsung Electronics Co., Ltd.
    Inventors: Ho-sang Sung, Eun-mi Oh
  • Patent number: 8977543
    Abstract: A quantizing apparatus is provided that includes a quantization path determiner that determines a path from a first path not using inter-frame prediction and a second path using the inter-frame prediction, as a quantization path of an input signal, based on a criterion before quantization of the input signal; a first quantizer that quantizes the input signal, if the first path is determined as the quantization path of the input signal; and a second quantizer that quantizes the input signal, if the second path is determined as the quantization path of the input signal.
    Type: Grant
    Filed: April 23, 2012
    Date of Patent: March 10, 2015
    Assignee: Samsung Electronics Co., Ltd.
    Inventors: Ho-sang Sung, Eun-mi Oh