Voiced Or Unvoiced Patents (Class 704/214)
  • Patent number: 10431233
    Abstract: Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S1 to a sampling rate S2. A power spectrum of a LP synthesis filter is computed, at the sampling rate S1, using the LP filter parameters. The power spectrum of the LP synthesis filter is modified to convert it from the sampling rate S1 to the sampling rate S2. The modified power spectrum of the LP synthesis filter is inverse transformed to determine autocorrelations of the LP synthesis filter at the sampling rate S2. The autocorrelations are used to compute the LP filter parameters at the sampling rate S2.
    Type: Grant
    Filed: November 15, 2017
    Date of Patent: October 1, 2019
    Assignee: VoiceAge EVS LLC
    Inventors: Redwan Salami, Vaclav Eksler
  • Patent number: 10354671
    Abstract: A voice coder configured to resolve periodic and aperiodic components of spectra is disclosed. The method of voice coding includes parsing the speech signal into a plurality of speech frames; for each of the plurality of speech frames: (a) generating the spectra for the speech frame, (b) parsing the spectra of the speech frame into a plurality of sub-bands, (c) transforming each of the plurality of sub-bands into a time-domain envelope signal, and (d) generating a plurality of sub-band voicing factors, wherein each sub-band voicing factor indicates the harmonicity of one of the plurality of sub-bands, and each sub-band voicing factor is based on the periodicity of one of said time-domain envelope signals associated with one of the plurality of sub-bands.
    Type: Grant
    Filed: February 21, 2018
    Date of Patent: July 16, 2019
    Assignee: OBEN, INC.
    Inventors: Kantapon Kaewtip, Fernando Villavicencio, Mark Harvilla
  • Patent number: 10332543
    Abstract: Example systems and methods capture a first plurality of portions of audio data by periodically capturing the audio data at first intervals. Embodiments detect speech onset in the audio data. Responsive to detection of the speech onset, systems and methods switch from periodically capturing the audio data to continuously capturing the audio data. Embodiments combine at least one captured portion of the first plurality of captured portions of the audio data with the continuously captured audio data to provide contiguous audio data.
    Type: Grant
    Filed: June 22, 2018
    Date of Patent: June 25, 2019
    Assignee: Cypress Semiconductor Corporation
    Inventors: Robert Zopf, Victor Simileysky, Ashutosh Pandey, Patrick Cruise
  • Patent number: 10163453
    Abstract: An electronic device or method for adjusting a gain on a voice operated control system can include one or more processors and a memory having computer instructions. The instructions, when executed by the one or more processors causes the one or more processors to perform the operations of receiving a first microphone signal, receiving a second microphone signal, updating a slow time weighted ratio of the filtered first and second signals, and updating a fast time weighted ratio of the filtered first and second signals. The one or more processors can further perform the operations of calculating an absolute difference between the fast time weighted ratio and the slow time weighted ratio, comparing the absolute difference with a threshold, and increasing the gain when the absolute difference is greater than the threshold. Other embodiments are disclosed.
    Type: Grant
    Filed: October 26, 2015
    Date of Patent: December 25, 2018
    Assignee: Staton Techiya, LLC
    Inventor: John Usher
  • Patent number: 10146868
    Abstract: Apparatuses, systems, methods, and media for filtering a data stream are provided. The data stream is partitioned into a plurality of data stream segments. An acoustic parameter is measured in each of the data stream segments. It is determined whether the acoustic parameter satisfies a first predetermined condition. The first predetermined condition includes a number of variances, in which the acoustic parameter exceeds a predetermined variance threshold, exceeding a predetermined number threshold. An extraneous portion of the data stream is identified in which the first predetermined condition is satisfied. It is determined whether the extraneous portion satisfies a second predetermined condition in the data stream. The extraneous portion is deleted from the data stream to produce a filtered data stream in response to the second predetermined condition being satisfied.
    Type: Grant
    Filed: June 8, 2017
    Date of Patent: December 4, 2018
    Assignee: AT&T INTELLECTUAL PROPERTY I, L.P.
    Inventors: Yeon-Jun Kim, I. Dan Melamed, Bernard S. Renger, Steven Neil Tischer
  • Patent number: 10140305
    Abstract: Systems and methods for structuring information include determining information quantity (IQ) and information value (IV) in an original digital information file (ODIF). An initial manipulation process applied to the ODIF forms a first resulting DIF (FRDIF), and a subsequent manipulation process applied to the FRDIF forms a second resulting DIF, wherein each manipulation process removes at least one element of the processed DIF and/or represents an element combination with a representative element and a first indicia of an interrelationship between the representative element and one or more elements in the combination, to reduce the IQ of the processed DIF, while retaining the IV thereof within a threshold. Manipulation processes are successively applied to the previously resulting DIF until successive applications do not achieve a threshold reduction in IQ. The last resulting DIF has a primary structure with a reduced IQ and an IV within the threshold of the original IV.
    Type: Grant
    Filed: October 14, 2016
    Date of Patent: November 27, 2018
    Assignee: GENERAL HARMONICS INTERNATIONAL INC.
    Inventors: Alexander Zhirkov, Alexey Oraevsky, Andrei Grichine, George Blondheim, Max Wandinger, Wade Attwood
  • Patent number: 10079023
    Abstract: A comfort noise generation apparatus constituted of: near and far end speech detectors arranged to detect speech activity in near-end and far-end signals and a comfort noise generator, wherein responsive to an indication from the near-end speech detector that speech activity is absent on the near-end signal and an indication from the far-end silence detector that speech activity is absent on the far-end signal, the comfort noise generator is arranged to initiate a determination of an estimation of near-end background noise, wherein responsive to an indication from the near-end speech detector that speech activity is present on the near-end signal or an indication from the far-end silence detector that speech activity is present on the far-end signal, the comfort noise generator is arranged to terminate the estimation determination of near-end background noise, and wherein the comfort noise generator is arranged to output a function of the near-end background noise estimation.
    Type: Grant
    Filed: September 22, 2016
    Date of Patent: September 18, 2018
    Assignee: Microsemi Semiconductor (U.S.) Inc.
    Inventors: Tanmay Zargar, Dillon Reed Ritter, Rodolfo Silva
  • Patent number: 10068580
    Abstract: An oversampling LPF unit receives a sound signal. A differentiator differentiates the sound signal. An overtone computation unit generates an overtone signal by multiplying a signal differentiated by the differentiator by the sound signal from the oversampling LPF unit. A HPF unit filters the overtone signal generated by the overtone computation unit. A combiner combines the overtone signal filtered by the HPF unit and the sound signal from the oversampling LPF unit.
    Type: Grant
    Filed: December 23, 2016
    Date of Patent: September 4, 2018
    Assignee: JVC KENWOOD Corporation
    Inventor: Tatsuya Onoda
  • Patent number: 10043534
    Abstract: A method and device for automatically increasing the spectral bandwidth of an audio signal including generating a “mapping” (or “prediction”) matrix based on the analysis of a reference wideband signal and a reference narrowband signal, the mapping matrix being a transformation matrix to predict high frequency energy from a low frequency energy envelope, generating an energy envelope analysis of an input narrowband audio signal, generating a resynthesized noise signal by processing a random noise signal with the mapping matrix and the envelope analysis, high-pass filtering the resynthesized noise signal, and summing the high-pass filtered resynthesized noise signal with the original an input narrowband audio signal. Other embodiments are disclosed.
    Type: Grant
    Filed: December 22, 2014
    Date of Patent: August 7, 2018
    Assignee: Staton Techiya, LLC
    Inventors: John Usher, Dan Ellis
  • Patent number: 9837089
    Abstract: A device for signal processing includes a receiver and a high-band excitation signal generator. The receiver is configured to receive a parameter associated with a bandwidth-extended audio stream. The high-band excitation signal generator is configured to determine a value of the parameter. The high-band excitation signal generator is also configured to select, based on the value of the parameter, one of target gain information associated with the bandwidth-extended audio stream or filter information associated with the bandwidth-extended audio stream. The high-band excitation signal generator is further configured to generate a high-band excitation signal based on the one of the target gain information or the filter information.
    Type: Grant
    Filed: May 25, 2016
    Date of Patent: December 5, 2017
    Assignee: QUALCOMM Incorporated
    Inventors: Venkatraman Atti, Venkata Subrahmanyam Chandra Sekhar Chebiyyam
  • Patent number: 9767791
    Abstract: Method and apparatus for segmenting speech by detecting the pauses between the words and/or phrases, and to determine whether a particular time interval contains speech or non-speech, such as a pause.
    Type: Grant
    Filed: September 12, 2016
    Date of Patent: September 19, 2017
    Assignee: SPEECH MORPHING SYSTEMS, INC.
    Inventors: Fathy Yassa, Ben Reaves, Nima Ferdosi
  • Patent number: 9711158
    Abstract: An encoding technique encoding a sound signal at a low bit rate with reduced processing. The technique includes: an interval determination determining an interval T between samples corresponding to periodicity of an audio signal or an integer multiple of a fundamental frequency of the audio signal from a set S of candidates for the interval T; and a side information generating encoding the determined interval T to obtain side information. The interval determining determines the interval T from a set S of Y candidates (Y<Z) including Z2 candidates (Z2<Z) selected from among Z candidates for the interval T representable with the side information without depending on a candidate subjected to the interval determination in a previous frame a predetermined number of frames before the current frame and including a candidate subjected to the interval determination in the previous frame the predetermined number of frames before the current frame.
    Type: Grant
    Filed: January 18, 2012
    Date of Patent: July 18, 2017
    Assignee: NIPPON TELEGRAPH AND TELEPHONE CORPORATION
    Inventors: Takehiro Moriya, Noboru Harada, Yusuke Hiwasaki, Yutaka Kamamoto
  • Patent number: 9703865
    Abstract: Apparatuses, systems, methods, and media for filtering a data stream are provided. The data stream is analyzed based on an acoustic parameter to determine extraneous portions in which a first predetermined condition is satisfied. When a first extraneous portion is separated from a second extraneous portion by a non-extraneous portion in which the first predetermined condition is not satisfied, it is determined whether the first extraneous portion being separated from the second extraneous portion by the non-extraneous portion satisfies a second predetermined condition. At least one of the first extraneous portion and the second extraneous portion is deleted from the data stream to produce a filtered data stream in response to determining the second predetermined condition is satisfied.
    Type: Grant
    Filed: September 25, 2015
    Date of Patent: July 11, 2017
    Assignee: AT&T INTELLECTUAL PROPERTY I, L.P.
    Inventors: Yeon-Jun Kim, I. Dan Melamed, Bernard S. Renger, Steven Neil Tischer
  • Patent number: 9679577
    Abstract: A voice switching device includes a learning unit configured to learn a background noise model expressing background noise contained in a first voice signal, based on the first voice signal, while the first voice signal having a first frequency band is received; a pseudo noise generation unit configured to generate pseudo noise expressing noise in a pseudo manner, based on the background noise model, after a first time point when the first voice signal is last received in a case where a received voice signal is switched from the first voice signal to a second voice signal having a second frequency band narrower than the first frequency band; and a superimposing unit configured to superimpose the pseudo noise on the second voice signal after the first time point.
    Type: Grant
    Filed: July 15, 2015
    Date of Patent: June 13, 2017
    Assignee: FUJITSU LIMITED
    Inventor: Kaori Endo
  • Patent number: 9583114
    Abstract: The invention provides an audio decoder being configured for decoding a bitstream so as to produce therefrom an audio output signal, the bitstream including at least an active phase followed by at least an inactive phase, wherein the bitstream has encoded therein at least a silence insertion descriptor frame which describes a spectrum of a background noise, the audio decoder including: a silence insertion descriptor decoder configured to decode the silence insertion descriptor frame; a decoding device configured to reconstruct the audio output signal from the bitstream during the active phase; a spectral converter configured to determine a spectrum of the audio output signal; a noise estimator device configured to determine a first spectrum of the noise of the audio output signal; a resolution converter configured to establish a second spectrum of the noise of the audio output signal; a comfort noise spectrum estimation device; and a comfort noise generator.
    Type: Grant
    Filed: June 19, 2015
    Date of Patent: February 28, 2017
    Assignee: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
    Inventors: Anthony Lombard, Martin Dietz, Stephan Wilde, Emmanuel Ravelli, Panji Setiawan, Markus Multrus
  • Patent number: 9576593
    Abstract: Techniques are described for calculating one or more verbal fluency scores for a person. An example method includes classifying, by a computing device, samples of audio data of speech of a person, based on amplitudes of the samples, into a first class of samples including speech or sound and a second class of samples including silence. The method further includes analyzing the first class of samples to determine a number of words spoken by the person, and calculating a verbal fluency score for the person based at least in part on the determined number of words spoken by the person.
    Type: Grant
    Filed: March 14, 2013
    Date of Patent: February 21, 2017
    Assignee: Regents of the University of Minnesota
    Inventors: Serguei V. S. Pakhomov, Laura Sue Hemmy, Kelvin O. Lim
  • Patent number: 9542358
    Abstract: An electromagnetic interference (EMI) signal is processed by digitizing the EMI signal, generating a plurality of overlapping time records from the digitized EMI signal, applying a window function to the plurality of overlapping time records to produce a plurality of modified time records, wherein the window function has a substantially flat top, and performing a fast Fourier transform (FFT) on each of the modified time records to produce a plurality of corresponding amplitude envelopes.
    Type: Grant
    Filed: August 16, 2013
    Date of Patent: January 10, 2017
    Assignee: Keysight Technologies, Inc.
    Inventors: Joseph M. Gorin, Michael E. Barnard
  • Patent number: 9501494
    Abstract: Systems and methods for structuring information include determining information quantity (IQ) and information value (IV) in an original digital information file (ODIF). An initial manipulation process applied to the ODIF forms a first resulting DIF (FRDIF), and a subsequent manipulation process applied to the FRDIF forms a second resulting DIF, wherein each manipulation process removes at least one element of the processed DIF and/or represents an element combination with a representative element and a first indicia of an interrelationship between the representative element and one or more elements in the combination, to reduce the IQ of the processed DIF, while retaining the IV thereof within a threshold. Manipulation processes are successively applied to the previously resulting DIF until successive applications do not achieve a threshold reduction in IQ. The last resulting DIF has a primary structure with a reduced IQ and an IV within the threshold of the original IV.
    Type: Grant
    Filed: February 18, 2014
    Date of Patent: November 22, 2016
    Assignee: General Harmonics International, Inc.
    Inventors: Alexander Zhirkov, Alexey Oraevsky, Andrei Grichine, George Blondheim, Max Wandinger, Wade Attwood
  • Patent number: 9390725
    Abstract: The present disclosure describes a system (100) for reducing background noise from a speech audio signal generated by a user. The system (100) includes a user device (102) receiving the speech audio signal, a noise reduction device (118) in communication with a stored data repository (208), where the noise reduction device is configured to convert the speech audio signal to text; generate synthetic speech based on the converted text; optionally determine the user as an actual subscriber based on a comparison between the speech audio signal with the synthetic speech; and selectively transmit the speech audio signal or the synthetic speech based on comparison between the predicted subjective quality of the recorded speech and the synthetic speech.
    Type: Grant
    Filed: July 1, 2015
    Date of Patent: July 12, 2016
    Assignee: ClearOne Inc.
    Inventor: Derek Graham
  • Patent number: 9373342
    Abstract: The present disclosure is directed towards a method for speech intelligibility. The method may include receiving, at one or more computing devices, a first speech input from a first user and performing voice activity detection upon the first speech input. The method may also include analyzing a spectral tilt associated with the first speech input, wherein analyzing includes computing an impulse response of a linear predictive coding (“LPC”) synthesis filter in a linear pulse code modulation (“PCM”) domain and wherein the one or more computing devices includes an adaptive high pass filter configured to recalculate one or more linear prediction coefficients.
    Type: Grant
    Filed: June 23, 2014
    Date of Patent: June 21, 2016
    Assignee: Nuance Communications, Inc.
    Inventors: Sridhar Pilli, Mahesh Godavarti, Qian-Yu Tang, Jose Lainez, Jagadeesh Balam
  • Patent number: 9330672
    Abstract: A frame loss compensation method and apparatus for audio signals are disclosed. The method includes: when a first frame immediately following a correctly received frame is lost, judging a frame type of the first lost frame, and when the first lost frame is a non-multi-harmonic frame, calculating MDCT coefficients of the first lost frame by using MDCT coefficients of one or more frames prior to the first lost frame; obtaining an initially compensated signal of the first lost frame according to the MDCT coefficients of the first lost frame; and performing a first class of waveform adjustment on the initially compensated signal of the first lost frame and taking an adjusted time-domain signal as a time-domain signal of the first lost frame. The apparatus includes a frame type judgment module, an MDCT coefficient acquisition module, an initial compensation signal acquisition module and an adjustment module.
    Type: Grant
    Filed: September 29, 2012
    Date of Patent: May 3, 2016
    Assignee: ZTE Corporation
    Inventors: Xu Guan, Hao Yuan, Ke Peng, Jiali Li
  • Patent number: 9299363
    Abstract: A time warp contour calculator for use in an audio signal decoder receives an encoded warp ratio information, derives a sequence of warp ratio values from the encoded warp ratio information, and obtains warp contour node values starting from a time warp contour start value. Ratios between the time warp contour node values and the time warp contour starting value are determined by the warp ratio values. The time warp contour calculator computes a time warp contour node value of a given time warp contour node, on the basis of a product-formation having a ratio between the time warp contour node values of the intermediate time warp contour node and the time warp contour starting value and a ratio between the time warp contour node values of the given time warp contour node and of the intermediate time warp contour node as factors.
    Type: Grant
    Filed: July 1, 2009
    Date of Patent: March 29, 2016
    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: 9275650
    Abstract: A new hybrid audio decoder and a new hybrid audio encoder having block switching for speech signals and audio signals are provided. Currently, very low bitrate audio coding methods for speech and audio signals are proposed. These audio coding methods cause very long delays. Generally, in coding an audio signal, an algorithm delay tends to be long to achieve higher frequency resolution. In coding a speech signal, the delay needs to be reduced because the speech signal is used for telecommunication. To balance fine coding quality for speech and audio input signals with very low bitrate, a combination of a low delay filter bank like AAC-ELD and a CELP coding method is provided.
    Type: Grant
    Filed: June 14, 2011
    Date of Patent: March 1, 2016
    Assignee: PANASONIC CORPORATION
    Inventors: Tomokazu Ishikawa, Takeshi Norimatsu, Haishan Zhong, Kok Seng Chong, Huan Zhou
  • Patent number: 9245539
    Abstract: This invention provides a voiced sound interval detection device which enables appropriate detection of a voiced sound interval of an observation signal even when a volume of sound from a sound source varies or when the number of sound sources is unknown or when different kinds of microphones are used together.
    Type: Grant
    Filed: January 25, 2012
    Date of Patent: January 26, 2016
    Assignee: NEC CORPORATION
    Inventor: Yoshifumi Onishi
  • Patent number: 9240184
    Abstract: A method and system for frame-level merging of HMM state predictions determined by different techniques is disclosed. An audio input signal may be transformed into a first and second sequence of feature vector, the sequences corresponding to each other and to a temporal sequence of frames of the audio input signal on a frame-by-frame basis. The first sequence may be processed by a neural network (NN) to determine NN-based state predictions, and the second sequence may be processed by a Gaussian mixture model (GMM) to determine GMM-based state predictions. The NN-based and GMM-based state predictions may be merged as weighted sums for each of a plurality of HMM state on a frame-by-frame basis to determine merged state predictions. The merged state predictions may then be applied to the HMMs to speech content of the audio input signal.
    Type: Grant
    Filed: February 12, 2013
    Date of Patent: January 19, 2016
    Assignee: Google Inc.
    Inventors: Hui Lin, Xin Lei, Vincent Vanhoucke
  • Patent number: 9183177
    Abstract: Methods, apparatuses, and media for filtering a data stream are provided. The data stream is partitioned into a plurality of data stream segments. An acoustic parameter of each of the data stream segments is measured, and it is determined whether the acoustic parameter of each of the data stream segments satisfies a predetermined condition. Extraneous segments of the data stream segments are identified in which the predetermined condition is satisfied, and it is determined whether the extraneous segments have a predetermined relationship in the data stream. The extraneous segments are deleted from the data stream to produce a filtered data stream in response to the extraneous segments having the predetermined relationship.
    Type: Grant
    Filed: April 22, 2013
    Date of Patent: November 10, 2015
    Assignee: AT&T INTELLECTUAL PROPERTY I, L.P.
    Inventors: Yeon-Jun Kim, I. Dan Melamed, Steven Neil Tischer, Bernard S. Renger
  • Patent number: 9135809
    Abstract: A remote control device includes a digital audio storage device, a talk button, and an optical distance measurer. The digital audio storage device is configured to continually record an audio input for a specific amount of time. The talk button is coupled to the digital audio storage device and is configured to initiate a transmission of the audio input to a set-top box device. The optical distance measurer is coupled to the talk button and is configured to automatically measure a distance to a user in response to the talk button being pressed.
    Type: Grant
    Filed: June 20, 2008
    Date of Patent: September 15, 2015
    Assignee: AT&T Intellectual Property I, LP
    Inventors: Hisao M. Chang, Iker Arizmendi
  • Patent number: 9070375
    Abstract: A voice activity detection method in a low SNR environment. The voice activity detection is performed by extracting a long-term spectrum variation component and a harmonic structure as feature vectors from a speech signal and increasing difference in feature vectors between speech and non-speech (i) using the long-term spectrum variation component feature or (ii) using a long-term spectrum variation component extraction and a harmonic structure feature extraction. A correct rate and an accuracy rate of the voice activity detection is improved over conventional methods by using a long-term spectrum variation component having a window length over an average phoneme duration of an utterance in the speech signal. The voice activity detection system and method provides speech processing, automatic speech recognition, and speech output capable of very accurate voice activity detection.
    Type: Grant
    Filed: February 27, 2009
    Date of Patent: June 30, 2015
    Assignee: International BUsiness Machines Corporation
    Inventors: Takashi Fukuda, Osamu Ichikawa, Masafumi Nishimura
  • Patent number: 9009034
    Abstract: A Voice Activity Detection/Silence Suppression (VAD/SS) system is connected to a channel of a transmission pipe. The channel provides a pathway for the transmission of energy. A method for operating a VAD/SS system includes detecting the energy on the channel, and activating or suppressing activation of the VAD/SS system depending upon the nature of the energy detected on the channel.
    Type: Grant
    Filed: November 12, 2014
    Date of Patent: April 14, 2015
    Assignee: AT&T Intellectual Property II, L.P.
    Inventors: Bing Chen, James H. James
  • Patent number: 9009054
    Abstract: This invention relates to retrieval for multimedia content, and provides a program endpoint time detection apparatus for detecting an endpoint time of a program by performing processing on audio signals of said program, comprising an audio classification unit for classifying said audio signals into a speech signal portion and a non-speech signal portion; a keyword retrieval unit for retrieving, as a candidate endpoint keyword, an endpoint keyword indicating start or end of the program from said speech signal portion; a content analysis unit for performing content analysis on context of the candidate endpoint keyword retrieved by the keyword retrieval unit to determine whether the candidate endpoint keyword is a valid endpoint keyword; and a program endpoint time determination unit for performing statistics analysis based on the retrieval result of said keyword retrieval unit and the determination result of said content analysis unit, and determining the endpoint time of the program.
    Type: Grant
    Filed: October 28, 2010
    Date of Patent: April 14, 2015
    Assignees: Sony Corporation, Institute of Acoustics, Chinese Academy of Sciences
    Inventors: Kun Liu, Weiguo Wu, Li Lu, Qingwei Zhao, Yonghong Yan, Hongbin Suo
  • Patent number: 8996389
    Abstract: Various techniques are disclosed for reducing artifacts generated by time compression. by adapting the time compression based on the state of the received audio. The amount of time compression may be bounded based on audio characteristics. Another feature provides a way of determining the most correlated portions of segments of audio. Voiced speech may be distinguished from unvoiced speech. Another feature provides a way of distinguishing between silence, voiced speech, and unvoiced speech. Time compression may be adapted during periods of lengthy silence. Another feature allows for reducing time compression during sensitive portions of the received audio. One or more of these features may be present in different embodiments.
    Type: Grant
    Filed: June 14, 2011
    Date of Patent: March 31, 2015
    Assignee: Polycom, Inc.
    Inventor: Eric David Elias
  • Patent number: 8996380
    Abstract: Systems and methods of synchronizing media are provided. A client device may be used to capture a sample of a media stream being rendered by a media rendering source. The client device sends the sample to a position identification module to determine a time offset indicating a position in the media stream corresponding to the sampling time of the sample, and optionally a timescale ratio indicating a speed at which the media stream is being rendered by the media rendering source based on a reference speed of the media stream. The client device calculates a real-time offset using a present time, a timestamp of the media sample, the time offset, and optionally the timescale ratio. The client device then renders a second media stream at a position corresponding to the real-time offset to be in synchrony to the media stream being rendered by the media rendering source.
    Type: Grant
    Filed: May 4, 2011
    Date of Patent: March 31, 2015
    Assignee: Shazam Entertainment Ltd.
    Inventors: Avery Li-Chun Wang, Rahul Powar, William Michael Mills, Christopher Jacques Penrose Barton, Philip Georges Inghelbrecht, Dheeraj Shankar Mukherjee
  • Patent number: 8990094
    Abstract: An electronic device for coding a transient frame is described. The electronic device includes a processor and executable instructions stored in memory that is in electronic communication with the processor. The electronic device obtains a current transient frame. The electronic device also obtains a residual signal based on the current transient frame. Additionally, the electronic device determines a set of peak locations based on the residual signal. The electronic device further determines whether to use a first coding mode or a second coding mode for coding the current transient frame based on at least the set of peak locations. The electronic device also synthesizes an excitation based on the first coding mode if the first coding mode is determined. The electronic device also synthesizes an excitation based on the second coding mode if the second coding mode is determined.
    Type: Grant
    Filed: September 8, 2011
    Date of Patent: March 24, 2015
    Assignee: QUALCOMM Incorporated
    Inventors: Venkatesh Krishnan, Ananthapadmanabhan Arasanipalai Kandhadai
  • Patent number: 8990079
    Abstract: When a voice-activated device or application is first started, the signal levels corresponding to spoken commands are initially unknown, making it difficult to set detection thresholds. The inventive method provides an initial command-detection threshold based on the noise level alone. The first command is detected using this initial threshold. Then the threshold is revised according to the first command sound, and a second command is detected using the revised threshold. After detecting each command, the detection threshold is further refined according to the current noise and command sounds. Methods are also disclosed for optimizing the thresholds, adjusting parameters according to sound, and detecting voiced and unvoiced sounds separately. These capabilities enable many emerging voice-activated products and applications.
    Type: Grant
    Filed: September 17, 2014
    Date of Patent: March 24, 2015
    Assignee: Zanavox
    Inventor: David Edward Newman
  • Patent number: 8982971
    Abstract: A multi-carrier signal is typically comprised of many equidistant sub-carriers. This results in periodicity of spectrum within the bandwidth of such a multi-carrier signal. An unknown multi-carrier signal with equidistant sub-carriers can thus be sensed together with its sub-carrier spacing by finding a discernable local maximum in the cepstrum (Fourier transform of the log spectrum) of the multi-carrier signal.
    Type: Grant
    Filed: March 29, 2012
    Date of Patent: March 17, 2015
    Assignee: QRC, Inc.
    Inventors: Sinisa Peric, Thomas F. Callahan, III
  • Patent number: 8984061
    Abstract: The conferencing system is composed of computers, a moderator's computer, and a projector connected on a network. The moderator's computer receives image data from the computers, and generates synthesized image data therefrom, which is transmitted to the projector for display of the synthesized image. The moderator's computer has the capability to switch the image being projected by the projector from the synthesized image to an image handled by one of the computers or by the moderator's computer. With such an arrangement, utilizing existing hardware resources it will be possible to display in a single split-screen display the images handled by the terminals connected on the network. Additionally, it will be possible to switch smoothly between on-screen displays, and to reduce the burden on the on-screen display operator in a networked conferencing system.
    Type: Grant
    Filed: August 6, 2008
    Date of Patent: March 17, 2015
    Assignee: Seiko Epson Corporation
    Inventor: Noboru Inoue
  • Publication number: 20150073783
    Abstract: In accordance with an embodiment of the present invention, a method for speech processing includes determining an unvoicing/voicing parameter reflecting a characteristic of unvoiced/voicing speech in a current frame of a speech signal comprising a plurality of frames. A smoothed unvoicing/voicing parameter is determined to include information of the unvoicing/voicing parameter in a frame prior to the current frame of the speech signal. A difference between the unvoicing/voicing parameter and the smoothed unvoicing/voicing parameter is computed. The method further includes generating an unvoiced/voiced decision point for determining whether the current frame comprises unvoiced speech or voiced speech using the computed difference as a decision parameter.
    Type: Application
    Filed: September 3, 2014
    Publication date: March 12, 2015
    Inventor: Yang Gao
  • Patent number: 8976906
    Abstract: A multi-carrier signal is typically comprised of many equidistant sub-carriers. This results in periodicity of spectrum within the bandwidth of such a multi-carrier signal. An unknown multi-carrier signal with equidistant sub-carriers can thus be sensed together with its sub-carrier spacing by finding a discernible local maximum in the cepstrum (Fourier transform of the log spectrum) of the multi-carrier signal.
    Type: Grant
    Filed: March 29, 2012
    Date of Patent: March 10, 2015
    Assignee: QRC, Inc.
    Inventors: Sinisa Peric, Thomas F. Callahan, III
  • Patent number: 8965773
    Abstract: A method is provided for hierarchical coding of a digital audio signal comprising, for a current frame of the input signal: a core coding, delivering a scalar quantization index for each sample of the current frame and at least one enhancement coding delivering indices of scalar quantization for each coded sample of an enhancement signal. The enhancement coding comprises a step of obtaining a filter for shaping the coding noise used to determine a target signal and in that the indices of scalar quantization of said enhancement signal are determined by minimizing the error between a set of possible values of scalar quantization and said target signal. The coding method can also comprise a shaping of the coding noise for the core bitrate coding. A coder implementing the coding method is also provided.
    Type: Grant
    Filed: November 17, 2009
    Date of Patent: February 24, 2015
    Assignee: Orange
    Inventors: Balazs Kovesi, Stéphane Ragot, Alain Le Guyader
  • Patent number: 8959025
    Abstract: Methods and systems for extracting speech from such packet streams. The methods and systems analyze the encoded speech in a given packet stream, and automatically identify the actual speech coding scheme that was used to produce it. These techniques may be used, for example, in interception systems where the identity of the actual speech coding scheme is sometimes unavailable or inaccessible. For instance, the identity of the actual speech coding scheme may be sent in a separate signaling stream that is not intercepted. As another example, the identity of the actual speech coding scheme may be sent in the same packet stream as the encoded speech, but in encrypted form.
    Type: Grant
    Filed: April 28, 2011
    Date of Patent: February 17, 2015
    Assignee: Verint Systems Ltd.
    Inventor: Genady Malinsky
  • Patent number: 8942975
    Abstract: Techniques are described herein that suppress noise in a Mel-filtered spectral domain. For example, a window may be applied to a representation of a speech signal in a time domain. The windowed representation in the time domain may be converted to a subsequent representation of the speech signal in the Mel-filtered spectral domain. A noise suppression operation may be performed with respect to the subsequent representation to provide noise-suppressed Mel coefficients.
    Type: Grant
    Filed: March 22, 2011
    Date of Patent: January 27, 2015
    Assignee: Broadcom Corporation
    Inventor: Jonas Borgstrom
  • Patent number: 8930183
    Abstract: A method of converting speech from the characteristics of a first voice to the characteristics of a second voice, the method comprising: receiving a speech input from a first voice, dividing said speech input into a plurality of frames; mapping the speech from the first voice to a second voice; and outputting the speech in the second voice, wherein mapping the speech from the first voice to the second voice comprises, deriving kernels demonstrating the similarity between speech features derived from the frames of the speech input from the first voice and stored frames of training data for said first voice, the training data corresponding to different text to that of the speech input and wherein the mapping step uses a plurality of kernels derived for each frame of input speech with a plurality of stored frames of training data of the first voice.
    Type: Grant
    Filed: August 25, 2011
    Date of Patent: January 6, 2015
    Assignee: Kabushiki Kaisha Toshiba
    Inventors: Byung Ha Chun, Mark John Francis Gales
  • Patent number: 8930184
    Abstract: A signal bandwidth extending apparatus including: a bandwidth extending section configured to extend a frequency bandwidth of a target signal, the target signal included in an input signal; a calculating section configured to calculate a degree of the target signal included in the input signal; and a controller configured to change a method of extending the frequency bandwidth by the bandwidth extending section according to a result of the calculating section.
    Type: Grant
    Filed: September 14, 2009
    Date of Patent: January 6, 2015
    Assignee: Kabushiki Kaisha Toshiba
    Inventors: Takashi Sudo, Masataka Osada
  • Patent number: 8924209
    Abstract: A method is disclosed for identifying a spoken command by detecting intervals of voiced and unvoiced sound, and then comparing the order of voiced and unvoiced sounds to a set of templates. Each template represents one of the predetermined acceptable commands of the application, and is associated with a predetermined action. When the order of voiced and unvoiced intervals in the spoken command matches the order in one of the templates, the associated action is thus selected. Silent intervals in the command may also be included for enhanced recognition. Efficient protocols are disclosed for discriminating voiced and unvoiced sounds, and for detecting the beginning and ending of each sound interval in the command, and for comparing the command sequence to the templates. In a sparse-command application, this method provides fast and robust recognition, and can be implemented with low-cost hardware and extremely minimal software.
    Type: Grant
    Filed: September 12, 2012
    Date of Patent: December 30, 2014
    Assignee: Zanavox
    Inventor: David Edward Newman
  • Patent number: 8924200
    Abstract: A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.
    Type: Grant
    Filed: September 28, 2011
    Date of Patent: December 30, 2014
    Assignee: Motorola Mobility LLC
    Inventors: Jonathan A. Gibbs, James P. Ashley, Udar Mittal
  • Patent number: 8924204
    Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.
    Type: Grant
    Filed: September 30, 2011
    Date of Patent: December 30, 2014
    Assignee: Broadcom Corporation
    Inventors: Juin-Hwey Chen, Jes Thyssen, Xianxian Zhang, Huaiyu Zeng
  • Patent number: 8909519
    Abstract: A Voice Activity Detection/Silence Suppression (VAD/SS) system is connected to a channel of a transmission pipe. The channel provides a pathway for the transmission of energy. A method for operating a VAD/SS system includes detecting the energy on the channel, and activating or suppressing activation of the VAD/SS system depending upon the nature of the energy detected on the channel.
    Type: Grant
    Filed: March 10, 2014
    Date of Patent: December 9, 2014
    Assignee: AT&T Intellectual Property II, L.P.
    Inventors: Bing Chen, James H. James
  • Patent number: 8898058
    Abstract: Systems, methods, apparatus, and machine-readable media for voice activity detection in a single-channel or multichannel audio signal are disclosed.
    Type: Grant
    Filed: October 24, 2011
    Date of Patent: November 25, 2014
    Assignee: QUALCOMM Incorporated
    Inventors: Jongwon Shin, Erik Visser, Ian Ernan Liu
  • Patent number: 8886527
    Abstract: A purpose is to suppress recognition process delay generated due to load in signal processing. Included is a speech input means 10 that inputs a speech signal, an output evaluation means 20 that evaluates whether or not the speech signal input by the speech input means 10 is the speech signal in a sound section, which is a speech section assuming that a speaker is speaking, and outputs the speech signal as a speech signal to be processed only when evaluated as the speech signal in the sound section, a signal processing means 30 that performs signal processing to the speech signal, which is output by the output evaluation means 20 as the speech signal to be processed, and a speech recognition processing means 40 that performs a speech recognition process to the speech signal which is signal-processed by the signal processing means 30.
    Type: Grant
    Filed: April 16, 2009
    Date of Patent: November 11, 2014
    Assignee: NEC Corporation
    Inventor: Toru Iwasawa
  • Patent number: 8880409
    Abstract: A system provided herein may perform automatic temporal alignment between music audio signal and lyrics with higher accuracy than ever. A non-fricative section extracting 4 extracts non-fricative sound sections, where no fricative sounds exist, from the music audio signal. An alignment portion 17 includes a phone model 15 for singing voice capable of estimating phonemes corresponding to temporal-alignment features. The alignment portion 17 performs an alignment operation using as inputs temporal-alignment features obtained from a temporal-alignment feature extracting portion 11, information on vocal and non-vocal sections obtained from a vocal section estimating portion 9, and a phoneme network SN on conditions that no phonemes exist at least in non-vocal sections and that no fricative phonemes exist in non-fricative sound sections.
    Type: Grant
    Filed: February 5, 2009
    Date of Patent: November 4, 2014
    Assignee: National Institute of Advanced Industrial Science and Technology
    Inventors: Hiromasa Fujihara, Masataka Goto