Excitation Patterns Patents (Class 704/223)
-
Patent number: 8805685Abstract: Disclosed herein are systems, methods, and tangible computer readable-media for detecting synthetic speaker verification. The method comprises receiving a plurality of speech samples of the same word or phrase for verification, comparing each of the plurality of speech samples to each other, denying verification if the plurality of speech samples demonstrate little variance over time or are the same, and verifying the plurality of speech samples if the plurality of speech samples demonstrates sufficient variance over time. One embodiment further adds that each of the plurality of speech samples is collected at different times or in different contexts. In other embodiments, variance is based on a pre-determined threshold or the threshold for variance is adjusted based on a need for authentication certainty. In another embodiment, if the initial comparison is inconclusive, additional speech samples are received.Type: GrantFiled: August 5, 2013Date of Patent: August 12, 2014Assignee: AT&T Intellectual Property I, L.P.Inventor: Horst J. Schroeter
-
Patent number: 8805681Abstract: A method and apparatus to search a codebook including pulses that model a predetermined component of a speech signal. The method includes the operations of selecting a predetermined number of paths corresponding to a predetermined number of pulse locations that are most consistent with the predetermined component, from among paths corresponding to pulse locations of a predetermined pulse location set allocated to at least one branch that connects one state of a predetermined Trellis structure to another state, performing the path selecting operation on each of states other than the one state, and selecting a path corresponding to pulse locations that are most consistent with the predetermined component, from among paths including the selected paths, wherein each path corresponds to a union of plural tracks of an Algebraic codebook. Accordingly, the number of calculations required during a codebook search is reduced.Type: GrantFiled: September 6, 2013Date of Patent: August 12, 2014Assignee: SAMSUNG Electronics Co., Ltd.Inventors: Hosang Sung, Kangeun Lee, Sang-won Kang, Thomas R. Fischer, Ja-kyoung Jun
-
Patent number: 8805695Abstract: A bandwidth expansion method and apparatus are disclosed, where the method includes: estimating a bandwidth of at least one decoded frame of a whole-band signal, so as to obtain an estimated bandwidth, where the estimated bandwidth corresponds to a whole-band signal that a decoded lower-band signal needs to be extended into; performing first predictive decoding on a part of the lower-band signal in a band above an effective bandwidth of the lower-band signal and below the estimated bandwidth, so as to obtain the part of the lower-band signal above the effective bandwidth of the lower-band signal and below the estimated bandwidth; and performing second predictive decoding on a part of the lower-band signal in a band above the estimated bandwidth, so as to obtain the part of the lower-band signal above the estimated bandwidth.Type: GrantFiled: July 22, 2013Date of Patent: August 12, 2014Assignee: Huawei Technologies Co., Ltd.Inventors: Zexin Liu, Lei Miao
-
Patent number: 8781821Abstract: A method is disclosed for controlling a voice-activated device by interpreting a spoken command as a series of voiced and non-voiced intervals. A responsive action is then performed according to the number of voiced intervals in the command. The method is well-suited to applications having a small number of specific voice-activated response functions. Applications using the inventive method offer numerous advantages over traditional speech recognition systems including speaker universality, language independence, no training or calibration needed, implementation with simple microcontrollers, and extremely low cost. For time-critical applications such as pulsers and measurement devices, where fast reaction is crucial to catch a transient event, the method provides near-instantaneous command response, yet versatile voice control.Type: GrantFiled: April 30, 2012Date of Patent: July 15, 2014Assignee: ZanavoxInventor: David Edward Newman
-
Patent number: 8768713Abstract: Systems and methods are disclosed for encoding audio in a set-top box that is invoked by a user when listening to a broadcast audio signal from a radio, TV, streaming or other audio device. A detection and identification system comprising an audio encoder is integrated in a set-top box, where detection and identification of media is realized. The encoding automatically identifies characteristics of the media (e.g., the source of a particular piece of material) by embedding an inaudible code within the content. This code contains information about the content that can be decoded by a machine, but is not detectable by human hearing. The embedded code may be used to provide programming information to the view or audience measurement date to the provider.Type: GrantFiled: March 15, 2010Date of Patent: July 1, 2014Assignee: The Nielsen Company (US), LLCInventors: Luc Chaoui, Taymoor Arshi, John Stavrapolous, Todd Cowling, Taher Behbehani
-
Patent number: 8768691Abstract: A sound encoder for efficiently encoding stereophonic sound. A prediction parameter analyzer determines a delay difference D and an amplitude ratio g of a first-channel sound signal with respect to a second-channel sound signal as channel-to-channel prediction parameters from a first-channel decoded signal and a second-channel sound signal. A prediction parameter quantizer quantizes the prediction parameters, and a signal predictor predicts a second-channel signal using the first decoded signal and the quantization prediction parameters. The prediction parameter quantizer encodes and quantizes the prediction parameters (the delay difference D and the amplitude ratio g) using a relationship (correlation) between the delay difference D and the amplitude ratio g attributed to a spatial characteristic (e.g., distance) from a sound source of the signal to a receiving point.Type: GrantFiled: March 23, 2006Date of Patent: July 1, 2014Assignee: Panasonic CorporationInventor: Koji Yoshida
-
Patent number: 8762136Abstract: The disclosure provides a speech encoder, decoder, speech processor and methods of encoding and decoding speech. In one embodiment, the speech encoder includes: (1) a speech frame generator configured to form a speech frame from an input speech signal, the speech frame having a length of multiple samples, (2) a speech frame processor configured to determine if the speech frame is a subsequent voiced frame of a group of consecutive voiced frames and, based thereon, perform speech analysis of the subsequent voiced frame; and (3) a speech frame coder configured to perform, if the speech frame is a subsequent voiced frame, differential coding of speech parameters of the subsequent voiced frame with respect to previous speech parameters of the previous voiced frame of the consecutive voiced frames.Type: GrantFiled: May 3, 2011Date of Patent: June 24, 2014Assignee: LSI CorporationInventors: Sooraj Kovoor Chathoth, Kumar U. Phani, Ganesh Guddanti
-
Patent number: 8762158Abstract: A method and apparatus for generating synthesis audio signals are provided. The method includes decoding a bitstream; splitting the decoded bitstream into n sub-band signals; generating n transformed sub-band signals by transforming the n sub-band signals in a frequency domain; and generating synthesis audio signals by respectively multiplying the n transformed sub-band signals by values corresponding to synthesis filter bank coefficients.Type: GrantFiled: August 5, 2011Date of Patent: June 24, 2014Assignee: Samsung Electronics Co., Ltd.Inventors: Hyun-wook Kim, Han-gil Moon, Sang-hoon Lee
-
Patent number: 8712771Abstract: The present invention relates to means and methods of automated difference recognition between speech and music signals in voice communication systems, devices, telephones, and methods, and more specifically, to systems, devices, and methods that automate control when either speech or music is detected over communication links. The present invention provides a novel system and method for monitoring the audio signal, analyze selected audio signal components, compare the results of analysis with a pre-determined threshold value, and classify the audio signal either as speech or music.Type: GrantFiled: October 31, 2013Date of Patent: April 29, 2014Inventor: Alon Konchitsky
-
Patent number: 8712765Abstract: A parameter decoding apparatus includes a prediction residue decoder that finds a quantized prediction residue based on encoded information included in a current frame subject to decoding and a moving-average predictor produces a predicted parameter by multiplying a predictive coefficient with a past quantized prediction residue. An adder decodes a parameter by adding the quantized prediction residue and the predicted parameter, wherein the prediction residue decoder, when the current frame is erased, finds a current-frame quantized prediction residue from a weighted linear sum of a parameter decoded in the past and a future-frame quantized prediction residue.Type: GrantFiled: May 17, 2013Date of Patent: April 29, 2014Assignee: Panasonic CorporationInventor: Hiroyuki Ehara
-
Patent number: 8712766Abstract: A method and system for analysis-by-synthesis encoding of an information signal is provide. The encoder (400) can include the steps of generating a first synthetic signal based on a first pitch-related codebook (402), generating a second synthetic signal based on a second pitch-related codebook (404), selecting a codebook configuration parameter based on the reference signal and the first and second synthetic signals, and conveying the codebook configuration for use in reconstructing an estimate of the input signal. The encoder can include an error expression having an error bias (506) and a prediction gain having a prediction gain bias (508) for determining the codebook configuration. The encoder can employ variable length coding and combinatorial subframe coding (600) for efficiently compressing the codebook configuration parameter and codebook related parameters for one or more subframes.Type: GrantFiled: May 16, 2006Date of Patent: April 29, 2014Assignee: Motorola Mobility LLCInventors: James P. Ashley, Udar Mittal
-
Patent number: 8706497Abstract: A synthesis filter 106 synthesizes a plurality of wide-band speech signals by combining wide-band phoneme signals and sound source signals from a speech signal code book 105, and a distortion evaluation unit 107 selects one of the wide-band speech signals with a minimum waveform distortion with respect to an up-sampled narrow-band speech signal output from a sampling conversion unit 101. A first bandpass filter 103 extracts a frequency component outside a narrow-band of the wide-band speech signal and a band synthesis unit 104 combines it with the up-sampled narrow-band speech signal.Type: GrantFiled: October 22, 2010Date of Patent: April 22, 2014Assignee: Mitsubishi Electric CorporationInventors: Satoru Furuta, Hirohisa Tasaki
-
Patent number: 8682656Abstract: A codebook generation system and associated methods are generally described herein. A method of generating a precoding matrix during closed-loop spatial multiplexing of wireless data may comprise selecting a vector from among vectors stored in a memory based on a received index, applying a Householder transformation to the selected vector to produce an output matrix, forming the precoding matrix from one or more columns of the output matrix, and precoding data for transmission over a wireless channel using the precoding matrix. Other embodiments are described and claimed.Type: GrantFiled: August 28, 2009Date of Patent: March 25, 2014Assignee: Intel CorporationInventors: Xintian E. Lin, Qinghua Li
-
Patent number: 8655651Abstract: The invention relates to a method, computer, computer program and computer program product for speech quality estimation. The method comprises the steps of: determining a coding distortion parameter (QCOD), a bandwidth related distortion parameter (BW) and a presentation level distortion parameter (PL) of a speech signal; extracting a first coefficient (?l) and a second coefficient (?2), the first coefficient and the second coefficient being dependent on the coding distortion parameter; and calculating a signal quality measure (Q), where the signal quality measure is QCOD+?1BW+?2PL using the signal quality measure in a quality estimation of the speech signal.Type: GrantFiled: July 26, 2010Date of Patent: February 18, 2014Assignee: Telefonaktiebolaget L M Ericsson (publ)Inventors: Volodya Grancharov, Mats Folkesson
-
Patent number: 8639294Abstract: A Dynamic Noise Compensation (DNC) telephone speech enhancement algorithm addresses the issue of environment noise on the listener end of a telephone call. A single microphone proximal to the listener provides a sample of near end ambient noise level and of near end speech. A Voice Activity Detector (VAD) detects the presence of near end (listener) speech. The DNC algorithm adjusts the far end incoming speech level based on the near end ambient noise and the VAD ensures that the near end listener speech does not effect the incoming speech level adjustment.Type: GrantFiled: May 1, 2012Date of Patent: January 28, 2014Assignee: Audyssey Laboratories, Inc.Inventors: Sunil Bharitkar, Nathan Dahlin, Ismael Hamad Nawfal
-
Patent number: 8635065Abstract: The present invention discloses an apparatus for automatic extraction of important events in audio signals comprising: signal input means for supplying audio signals; audio signal fragmenting means for partitioning audio signals supplied by the signal input means into audio fragments of a predetermined length and for allocating a sequence of one or more audio fragments to a respective audio window; feature extracting means for analyzing acoustic characteristics of the audio signals comprised in the audio fragments and for analyzing acoustic characteristics of the audio signals comprised in the audio windows; and important event extraction means for extracting important events in audio signals supplied by the audio signal fragmenting means based on predetermined important event classifying rules depending on acoustic characteristics of the audio signals comprised in the audio fragments and on acoustic characteristics of the audio signals comprised in the audio windows, wherein each important event extractedType: GrantFiled: November 10, 2004Date of Patent: January 21, 2014Assignee: Sony Deutschland GmbHInventors: Silke Goronzy-Thomae, Thomas Kemp, Ralf Kompe, Yin Hay Lam, Krzysztof Marasek, Raquel Tato
-
Patent number: 8630849Abstract: A method and apparatus to convert a linear predictive coding (LPC) coefficient into a coefficient having order characteristics, such as a line spectrum frequency (LSF), and to vector quantize the coefficient having the order characteristics when a speech signal is encoded. The method and apparatus split the vector of the coefficient having the order characteristics into a plurality of subvectors, select a codebook in which an available bit is variably allocated to each subvector according to distribution of elements of each subvector, and quantize each subvector according to the selected codebook. The method and apparatus use normalized codebooks.Type: GrantFiled: November 15, 2006Date of Patent: January 14, 2014Assignee: SAMSUNG Electronics Co., Ltd.Inventors: Chang-Yong Son, Eun-Mi Oh, Ho-Sang Sung, Kang-Eun Lee, Ki-Hyun Choo, Jung-Hoe Kim
-
Patent number: 8620645Abstract: A decoder arrangement comprising a receiver input for parameters of frame-based coded signals and a decoder arranged to provide frames of decoded audio signals based on the parameters. The receiver input and/or the decoder is arranged to establish a time difference between the occasion when parameters of a first frame is available at the receiver input and the occasion when a decoded audio signal of the first frame is available at an output of the decoder, which time difference corresponds to at least one frame. A postfilter is connected to the output of the decoder and to the receiver input. The postfilter is arranged to provide a filtering of the frames of decoded audio signals into an output signal in response to parameters of a respective subsequent frame.Type: GrantFiled: December 14, 2007Date of Patent: December 31, 2013Assignee: Telefonaktiebolaget L M Ericsson (publ)Inventor: Stefan Bruhn
-
Patent number: 8612532Abstract: A system and method disclosed for using and updating a database of template responses for a live agent in response to user communications. The method includes computing an average string distance between each response from a live agent and a template, use to generate the response, modifying the computed average string distance based on a customer satisfaction score associated with each response and selecting a response that minimizes the computed average string distance and maximizes customer satisfaction. Upon receiving a further communication on a certain issue, the system presents a prototype response that has been added to the template database to the live agent for use in generating a response to the further communication that reduces handling time and increases customer satisfaction.Type: GrantFiled: November 30, 2012Date of Patent: December 17, 2013Assignee: AT&T Intellectual Property I, L.P.Inventors: Srinivas Bangalore, Mazin Gilbert
-
Patent number: 8612215Abstract: A method and apparatus to extract an important frequency component of an audio signal and a method and apparatus to encode and/or decode an audio signal by using the same. The method of extracting an important frequency component of an audio signal includes converting an audio signal of a time domain into an audio signal of a frequency domain, selecting a frequency band having a harmonic feature from the converted audio signal of the frequency domain, and extracting an important frequency component from the selected frequency band having the harmonic feature.Type: GrantFiled: October 31, 2007Date of Patent: December 17, 2013Assignee: Samsung Electronics Co., Ltd.Inventors: Chang-yong Son, Bun-mi Oh, Ho-sang Sung, Ki-hyun Choo, Jung-hue Kim, Kang-eun Lee
-
Patent number: 8612216Abstract: To form an audio signal, frequency components of the audio signal which are allotted to a first subband are formed by means of a subband decoder using supplied fundamental period values which respectively indicate a fundamental period for the audio signal. Frequency components of the audio signal which are allotted to a second subband are formed by exciting an audio synthesis filter using an excitation signal which is specific to the second subband. To produce this excitation signal, an excitation signal generator derives a fundamental period parameter from the fundamental period values. The fundamental period parameter is used by the excitation signal generator to form pulses with a pulse shape which is dependent on the fundamental period parameter at an interval of time which is determined by the fundamental period parameter and to mix them with a noise signal.Type: GrantFiled: January 31, 2006Date of Patent: December 17, 2013Assignee: Siemens Enterprise Communications GmbH & Co. KGInventors: Martin Gartner, Bernd Geiser, Peter Jax, Stefan Schandl, Herve Taddei, Peter Vary
-
Patent number: 8600739Abstract: A coding method is adapted to select different codebook search algorithms according to varied types of input signals. An encoder using the coding method is also provided. As appropriate search algorithms may be selected according to all possible structural features of the input signals, certain types of signals for which satisfactory results may be obtained through simple computations may match with search algorithms suitable for these signal types and having low computation complexities, so as to achieve better performance with fewer system resources. Meanwhile, other types of signals that need complicated computations may be processed by more sophisticated search algorithms, thereby ensuring the coding quality.Type: GrantFiled: June 9, 2009Date of Patent: December 3, 2013Assignee: Huawei Technologies Co., Ltd.Inventors: Dejun Zhang, Liang Zhang, Yue Lang, Tinghong Wang, Lixiong Li, Wenhai Wu, Wei Xiao, Fuwei Ma, Zexin Liu
-
Patent number: 8600037Abstract: Echo cancellation is handled using a pattern classification technique. During a training phase, the communications device is trained to learn patterns of signal inputs that correspond to certain communication modes. After numerous different patterns have been classified by mode, the classified patterns are used during real-time use of the communications device in order to determine, dynamically, the mode in which the communications device is currently operating. The communications device may then apply, to the microphone-produced signal, a suppression action.Type: GrantFiled: June 1, 2012Date of Patent: December 3, 2013Assignee: Apple Inc.Inventor: Arvindh Krishnaswany
-
Patent number: 8595002Abstract: Encoding a sequence of digital speech samples into a bit stream includes dividing the digital speech samples into one or more frames, computing model parameters for a frame, and quantizing the model parameters to produce pitch bits conveying pitch information, voicing bits conveying voicing information, and gain bits conveying signal level information. One or more of the pitch bits are combined with one or more of the voicing bits and one or more of the gain bits to create a first parameter codeword that is encoded with an error control code to produce a first FEC codeword that is included in a bit stream for the frame. The process may be reversed to decode the bit stream.Type: GrantFiled: January 18, 2013Date of Patent: November 26, 2013Assignee: Digital Voice Systems, Inc.Inventor: John C. Hardwick
-
Patent number: 8595000Abstract: A method and an apparatus to encode and decode a speech signal using a code excited linear prediction (CELP) algorithm. In order to reduce a bit rate without degrading performance in an enhancement layer based on CELP, each of a fixed codebook of a core layer and a fixed codebook of the enhancement layer is divided into a plurality of spaces. The spaces of the fixed codebook of the enhancement layer excludes a space corresponding to a least distorted space determined from among the spaces of the fixed codebook of the core layer are searched.Type: GrantFiled: February 22, 2007Date of Patent: November 26, 2013Assignee: Samsung Electronics Co., Ltd.Inventors: Kangeun Lee, Eunmi Oh, Hosang Sung, Changyong Son, Kihyun Choo, Junghoe Kim
-
Patent number: 8589166Abstract: Systems and methods are described for performing packet loss concealment (PLC) to mitigate the effect of one or more lost frames within a series of frames that represent a speech signal. In accordance with the exemplary systems and methods, PLC is performed by searching a codebook of speech-related parameter profiles to identify content that is being spoken and by selecting a profile associated with the identified content for use in predicting or estimating speech-related parameter information associated with one or more lost frames of a speech signal. The predicted/estimated speech-related parameter information is then used to synthesize one or more frames to replace the lost frame(s) of the speech signal.Type: GrantFiled: September 21, 2010Date of Patent: November 19, 2013Assignee: Broadcom CorporationInventor: Robert W. Zopf
-
Patent number: 8571852Abstract: A scalable decoder device (50) for signals representing audio comprises a primary decoder (21) connected to an input (40). The primary decoder (21) is arranged to provide a primary decoded signal (23) based on received parameters (4). A primary postfilter (31) is connected to the primary decoder (23) to provide a primary postfiltered signal (32). A secondary enhancement decoder (45) is connected to the input (40) and arranged to provide a secondary decoded enhancement signal (44). The device further comprises a combiner arrangement (55), arranged for combining the primary postfiltered signal (32) and a signal (53) based on the secondary decoded enhancement signal (44) into an output signal (6) to be provided at an output (6). The combining is made with an adaptable strength relation between contributions from the two signals. A method for decoding coded signals representing audio operates in analogy with the scalable decoder device (50).Type: GrantFiled: December 14, 2007Date of Patent: October 29, 2013Assignee: Telefonaktiebolaget L M Ericsson (publ)Inventor: Stefan Bruhn
-
Patent number: 8571226Abstract: A sound reproducing device has a loudspeaker arranged to produce sound from an audio signal provided by an audio signal source. A microphone is positioned to pick up ambient noise and generate a microphone signal which comprises the noise. An ambient noise cancellation (ANC) system receives the microphone signal from the microphone and generates anti-noise corresponding to the ambient noise in the microphone signal. An automatic polarity adaptation (AAP) system monitors the ANC system and, when a decision criterion is fulfilled, causes a switch in polarity for the generated anti-noise.Type: GrantFiled: December 10, 2010Date of Patent: October 29, 2013Assignees: Sony Corporation, Sony Mobile Communications ABInventor: Peter Isberg
-
Patent number: 8566106Abstract: A method and device for searching an algebraic codebook during encoding of a sound signal, wherein the algebraic codebook comprises a set of codevectors formed of a number of pulse positions and a number of pulses distributed over the pulse positions. In the algebraic codebook searching method and device, a reference signal for use in searching the algebraic codebook is calculated. In a first stage, a position of a first pulse is determined in relation with the reference signal and among the number of pulse positions. In each of a number of stages subsequent to the first stage, (a) an algebraic codebook gain is recomputed, (b) the reference signal is updated using the recomputed algebraic codebook gain and (c) a position of another pulse is determined in relation with the updated reference signal and among the number of pulse positions.Type: GrantFiled: September 11, 2008Date of Patent: October 22, 2013Assignee: Voiceage CorporationInventors: Redwan Salami, Vaclav Eksler, Milan Jelinek
-
Patent number: 8560306Abstract: A method and apparatus to search a codebook including pulses that model a predetermined component of a speech signal. The method includes the operations of selecting a predetermined number of paths corresponding to a predetermined number of pulse locations that are most consistent with the predetermined component, from among paths corresponding to pulse locations of a predetermined pulse location set allocated to at least one branch that connects one state of a predetermined Trellis structure to another state, performing the path selecting operation on each of states other than the one state, and selecting a path corresponding to pulse locations that are most consistent with the predetermined component from among paths including the selected paths, wherein each path corresponds to a union of plural tracks of an algebraic codebook. Accordingly, a number of calculations required during a codebook search is reduced.Type: GrantFiled: July 13, 2006Date of Patent: October 15, 2013Assignee: Samsung Electronics Co., Ltd.Inventors: Hosang Sung, Kungeun Lee, Sang-won Kang, Thomas R. Fischer, Ja-kyoung Jun
-
Patent number: 8554548Abstract: An audio decoding device can adjust the high-range emphasis degree in accordance with a background noise level. The audio decoding device includes: a sound source signal decoder which performs a decoding process by using sound source encoding data separated by a separator so as to obtain a sound source signal; an LPC synthesis filter which performs an LPC synthesis filtering process by using a sound source signal and an LPC generated by an LPC decoder so as to obtain a decoded sound signal; a mode judger which determines whether a decoded sound signal is a stationary noise period by using a decoded LSP inputted from the LPC decoder a power calculator which calculates the power of the decoded audio signal; an SNR calculator which calculates an SNR of the decoded audio signal by using the power of the decoded audio signal and a mode judgment result in the mode judger and a post filter which performs a post filtering process by using the SNR of the decoded audio signal.Type: GrantFiled: February 29, 2008Date of Patent: October 8, 2013Assignee: Panasonic CorporationInventor: Hiroyuki Ehara
-
Patent number: 8554549Abstract: A voice encoding device accurately encodes a spectrum shape of a signal having a strong tonality such as a vowel. The device includes: a sub-band divider which divides a first layer error conversion coefficient to be encoded into M sub-bands so as to generate M sub-band conversion coefficients; a shape vector encoder which performs encoding on each of the M sub-band conversion coefficients so as to obtain M shape encoded information and calculates a target gain of each of the M sub-band conversion coefficients; a gain vector former which forms one gain vector by using M target gains; a gain vector encoder which encodes the gain vector so as to obtain gain encoded information; and a multiplexer which multiplexes the shape encoded information with the gain encoded information.Type: GrantFiled: February 29, 2008Date of Patent: October 8, 2013Assignee: Panasonic CorporationInventors: Masahiro Oshikiri, Toshiyuki Morii, Tomofumi Yamanashi
-
Patent number: 8542766Abstract: An apparatus and method for aligning input and feedback signals in a transmission circuit are provided. The method includes capturing an input signal and a feedback signal, determining a first time delay between the input signal and the feedback signal, determining a second time delay between the input signal and the feedback signal, the determination of the second time delay having a higher resolution than the determination of the first time delay, and applying the first time delay and the second time delay to temporally align the input signal with the feedback signal. Use of the present invention provides an improved resolution of time alignment while reducing the overall complexity and cost of the transmission circuit.Type: GrantFiled: April 29, 2011Date of Patent: September 24, 2013Assignee: Samsung Electronics Co., Ltd.Inventors: Igor Chekhovstov, Khalil Haddad
-
Patent number: 8538765Abstract: A parameter decoding apparatus includes a prediction residue decoder that finds a quantized prediction residue based on encoded information included in a current frame subject to decoding and an auto-regressive predictor produces a predicted parameter by multiplying a predictive coefficient with a past decoded parameter. An adder decodes a parameter by adding the quantized prediction residue and the predicted parameter, wherein the prediction residue decoder, when the current frame is erased, finds a current-frame quantized prediction residue from a weighted linear sum of a parameter decoded in the past and a future-frame quantized prediction residue.Type: GrantFiled: May 17, 2013Date of Patent: September 17, 2013Assignee: Panasonic CorporationInventor: Hiroyuki Ehara
-
Patent number: 8529473Abstract: A method and apparatus are provided for processing a set of communicated signals associated with a set of muscles, such as the muscles near the larynx of the person, or any other muscles the person use to achieve a desired response. The method includes the steps of attaching a single integrated sensor, for example, near the throat of the person proximate to the larynx and detecting an electrical signal through the sensor. The method further includes the steps of extracting features from the detected electrical signal and continuously transforming them into speech sounds without the need for further modulation. The method also includes comparing the extracted features to a set of prototype features and selecting a prototype feature of the set of prototype features providing a smallest relative difference.Type: GrantFiled: July 27, 2012Date of Patent: September 10, 2013Inventors: Michael Callahan, Thomas Coleman
-
Patent number: 8532982Abstract: A method and apparatus to encode and decode an audio/speech signal is provided. An inputted audio signal or speech signal may be transformed into at least one of a high frequency resolution signal and a high temporal resolution signal. The signal may be encoded by determining an appropriate resolution, the encoded signal may be decoded, and thus the audio signal, the speech signal, and a mixed signal of the audio signal and the speech signal may be processed.Type: GrantFiled: July 14, 2009Date of Patent: September 10, 2013Assignee: SAMSUNG Electronics Co., Ltd.Inventors: Eun Mi Oh, Jung Hoe Kim, Ki Hyun Choo, Ho Sang Sung, Mi Young Kim
-
Patent number: 8515767Abstract: Codebook indices for a scalable speech and audio codec may be efficiently encoded based on anticipated probability distributions for such codebook indices. A residual signal from a Code Excited Linear Prediction (CELP)-based encoding layer may be obtained, where the residual signal is a difference between an original audio signal and a reconstructed version of the original audio signal. The residual signal may be transformed at a Discrete Cosine Transform (DCT)-type transform layer to obtain a corresponding transform spectrum. The transform spectrum is divided into a plurality of spectral bands, where each spectral band having a plurality of spectral lines. A plurality of different codebooks are then selected for encoding the spectral bands, where each codebook is associated with a codebook index. A plurality of codebook indices associated with the selected codebooks are then encoded together to obtain a descriptor code that more compactly represents the codebook indices.Type: GrantFiled: November 3, 2008Date of Patent: August 20, 2013Assignee: QUALCOMM IncorporatedInventor: Yuriy Reznik
-
Publication number: 20130204615Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal.Type: ApplicationFiled: March 11, 2013Publication date: August 8, 2013Applicant: RESEARCH IN MOTION LIMITEDInventor: RESEARCH IN MOTION LIMITED
-
Patent number: 8504365Abstract: Disclosed herein are systems, methods, and tangible computer readable-media for detecting synthetic speaker verification. The method comprises receiving a plurality of speech samples of the same word or phrase for verification, comparing each of the plurality of speech samples to each other, denying verification if the plurality of speech samples demonstrate little variance over time or are the same, and verifying the plurality of speech samples if the plurality of speech samples demonstrates sufficient variance over time. One embodiment further adds that each of the plurality of speech samples is collected at different times or in different contexts. In other embodiments, variance is based on a pre-determined threshold or the threshold for variance is adjusted based on a need for authentication certainty. In another embodiment, if the initial comparison is inconclusive, additional speech samples are received.Type: GrantFiled: April 11, 2008Date of Patent: August 6, 2013Assignee: AT&T Intellectual Property I, L.P.Inventor: Horst Schroeter
-
Patent number: 8494863Abstract: The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises a linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; a quantization unit for quantizing a transform domain signal; a long term prediction unit for determining an estimation of the frame of the filtered input signal based on a reconstruction of a previous segment of the filtered input signal; and a transform domain signal combination unit for combining, in the transform domain, the long term prediction estimation and the transformed input signal to generate the transform domain signal.Type: GrantFiled: December 30, 2008Date of Patent: July 23, 2013Assignee: Dolby Laboratories Licensing CorporationInventors: Arijit Biswas, Heiko Purnhagen, Kristofer Kjoerling, Barbara Resch, Lars Villemoes, Per Hedelin
-
Patent number: 8494849Abstract: A method of transmitting speech data to a remote device in a distributed speech recognition system, includes the steps of: dividing an input speech signal into frames; calculating, for each frame, a voice activity value representative of the presence of speech activity in the frame; grouping the frames into multiframes, each multiframe including a predetermined number of frames; calculating, for each multiframe, a voice activity marker representative of the number of frames in the multiframe representing speech activity; and selectively transmitting, on the basis of the voice activity marker associated with each multiframe, the multiframes to the remote device.Type: GrantFiled: June 20, 2005Date of Patent: July 23, 2013Assignee: Telecom Italia S.p.A.Inventors: Ivano Salvatore Collotta, Donato Ettorre, Maurizio Fodrini, Pierluigi Gallo, Roberto Spagnolo
-
Patent number: 8489395Abstract: A method and an apparatus for generating a lattice vector quantizer codebook are disclosed. The method includes: storing an eigenvector set that includes amplitude vectors and/or length vectors, where the amplitude vectors and/or length vectors are different from each other and correspond to a root leader of a lattice vector quantizer; storing storage addresses of the amplitude vectors and length vectors, where the amplitude vectors and length vectors correspond to the root leader and are in the eigenvector set; and generating a lattice vector quantizer codebook according to the eigenvector set and the storage addresses.Type: GrantFiled: November 28, 2011Date of Patent: July 16, 2013Assignee: Huawei Technologies Co., Ltd.Inventors: Haiting Li, Deming Zhang
-
Patent number: 8489403Abstract: The APPARATUSES, METHODS AND SYSTEMS FOR SPARSE SINUSOIDAL AUDIO PROCESSING AND TRANSMISSION (hereinafter “SS-Audio”) provides a platform for encoding and decoding audio signals based on a sparse sinusoidal structure. In one embodiment, the SS-Audio encoder may encode received audio inputs based on its sparse representation in the frequency domain and transmit the encoded and quantized bit streams. In one embodiment, the SS-Audio decoder may decode received quantized bit streams based on sparse reconstruction and recover the original audio input by reconstructing the sinusoidal parameters in the frequency domain.Type: GrantFiled: August 25, 2010Date of Patent: July 16, 2013Assignee: Foundation For Research and Technology—Institute of Computer Science ‘FORTH-ICS’Inventors: Anthony Griffin, Athanasios Mouchtaris, Panagiotis Tsakalides
-
Patent number: 8489393Abstract: The perceived quality of a narrowband speech signal truncated from a wideband speech signal is improved by generating in a third frequency band third speech components matching first speech components in a first frequency band of the narrowband signal, and generating in a fourth frequency band fourth speech components matching second speech components in a second frequency band of the narrowband signal. A first gain factor is applied to the third speech components to generate adjusted third speech components, and a second gain factor is applied to the fourth speech components to generate adjusted fourth speech components, the gain factors being selected such that the ratios of the average powers of the adjusted third and fourth speech components to the average power of the first speech components are predetermined values.Type: GrantFiled: November 23, 2009Date of Patent: July 16, 2013Assignee: Cambridge Silicon Radio LimitedInventors: Rogerio Guedes Alves, Kuan-chieh Yen, Michael Christopher Vartanian, Sameer Arun Gadre
-
Patent number: 8484036Abstract: A wideband speech encoder according to one embodiment includes a narrowband encoder and a highband encoder. The narrowband encoder is configured to encode a narrowband portion of a wideband speech signal into a set of filter parameters and a corresponding encoded excitation signal. The highband encoder is configured to encode, according to a highband excitation signal, a highband portion of the wideband speech signal into a set of filter parameters. The highband encoder is configured to generate the highband excitation signal by applying a nonlinear function to a signal based on the encoded narrowband excitation signal to generate a spectrally extended signal.Type: GrantFiled: April 3, 2006Date of Patent: July 9, 2013Assignee: QUALCOMM IncorporatedInventors: Koen Bernard Vos, Ananthapadmanabhan Aasanipalai Kandhadai
-
Voice analysis device, voice analysis method, voice analysis program, and system integration circuit
Patent number: 8478587Abstract: A sound analysis device comprises: a sound parameter calculation unit operable to acquire an audio signal and calculate a sound parameter for each of partial audio signals, the partial audio signals each being the acquired audio signal in a unit of time; a category determination unit operable to determine, from among a plurality of environmental sound categories, which environmental sound category each of the partial audio signals belongs to, based on a corresponding one of the calculated sound parameters; a section setting unit operable to sequentially set judgement target sections on a time axis as time elapses, each of the judgment target sections including two or more of the units of time, the two or more of the units of time being consecutive; and an environment judgment unit operable to judge, based on a number of partial audio signals in each environmental sound category determined in at least a most recent judgment target section, an environment that surrounds the sound analysis device in at least theType: GrantFiled: March 13, 2008Date of Patent: July 2, 2013Assignee: Panasonic CorporationInventors: Takashi Kawamura, Ryouichi Kawanishi -
Patent number: 8473284Abstract: A voice encoding/decoding method and apparatus. A voice encoder includes: a quantization selection unit generating a quantization selection signal; and a quantization unit extracting a linear prediction coding (LPC) coefficient from an input signal, converting the extracted LPC coefficient into a line spectral frequency (LSF), quantizing the LSF with a first LSF quantization unit or a second LSF quantization unit based on the quantization selection signal, and converting the quantized LSF into a quantized LPC coefficient. The quantization selection signal selects the first LSF quantization unit or second LSF quantization unit based on characteristics of a synthesized voice signal in previous frames of the input signal.Type: GrantFiled: April 4, 2005Date of Patent: June 25, 2013Assignee: Samsung Electronics Co., Ltd.Inventors: Kangeun Lee, Hosang Sung, Kihyun Choo
-
Patent number: 8468015Abstract: A parameter decoding device performs a parameter compensation process so as to suppress degradation of a main observation quality in a prediction quantization. The parameter decoding device includes first amplifiers which multiply inputted quantization prediction residual vectors by a weighting coefficient. A further amplifier multiplies the preceding frame decoding LSF vector yn?1 by the weighting coefficient. An additional amplifier multiplies the code vector xn+1 outputted from a codebook by the weighting coefficient ?0. An adder calculates the total of the vectors outputted from the amplifiers, the further amplifier, and the additional amplifier. A selector switch selects the vector outputted from the adder if the frame erasure coding Bn of the current frame indicates that ‘the n-th frame is an erased frame’ and the frame erasure coding Bn+1 of the next frame indicates that ‘the n+1-th frame is a normal frame’.Type: GrantFiled: November 9, 2007Date of Patent: June 18, 2013Assignee: Panasonic CorporationInventor: Hiroyuki Ehara
-
Patent number: 8457962Abstract: This invention provides remote audio surveillance by recording audio data via three microphones and storage on a removable digital mass storage device, operating on battery power. The housing is of a weather resistant design to withstand outdoor conditions. Recording can be done in person or recording times can be defined so that the unit will only ‘listen’ during the desired times of the day, on a day to day basis. The user does not have to be in the vicinity but simply programs the record time(s) and leaves the device in the woods. The device also has play back capabilities for any recorded audio data and can interface with personal computers via the removable digital mass storage device. In addition to the audio collection and playback capabilities, PC software will be provided with the device which will analyze the data and provide direction of sound (based upon relative amplitude of the 3 microphones) and distance of sound (based on absolute and relative recorded amplitudes).Type: GrantFiled: August 4, 2006Date of Patent: June 4, 2013Inventor: Lawrence P. Jones
-
Patent number: 8452590Abstract: A fixed codebook searching apparatus, includes a convolution operator, implemented by at least one processor, that convolves an impulse response of a perceptually weighted synthesis filter with an impulse response vector that has values at negative times, to generate a second impulse response vector that has values at negative times. A matrix generator, implemented by at least one processor, generates a Toeplitz-type convolution matrix using the second impulse response vector generated by the convolution operator. A searcher, implemented by at least one processor, performs a codebook search by maximizing a term using the Toeplitz-type convolution matrix.Type: GrantFiled: April 25, 2011Date of Patent: May 28, 2013Assignee: Panasonic CorporationInventors: Hiroyuki Ehara, Koji Yoshida