Normalizing Patents (Class 704/224)
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Patent number: 8843367Abstract: An adaptive equalization system that adjusts the spectral shape of a speech signal based on an intelligibility measurement of the speech signal may improve the intelligibility of the output speech signal. Such an adaptive equalization system may include a speech intelligibility measurement module, a spectral shape adjustment module, and an adaptive equalization module. The speech intelligibility measurement module is configured to calculate a speech intelligibility measurement of a speech signal. The spectral shape adjustment module is configured to generate a weighted long-term speech curve based on a first predetermined long-term average speech curve, a second predetermined long-term average speech curve, and the speech intelligibility measurement. The adaptive equalization module is configured to adapt equalization coefficients for the speech signal based on the weighted long-term speech curve.Type: GrantFiled: May 4, 2012Date of Patent: September 23, 2014Assignee: 8758271 Canada Inc.Inventors: Phillip Alan Hetherington, Xueman Li
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APPARATUSES AND METHODS FOR MULTI-CHANNEL SIGNAL COMPRESSION DURING DESIRED VOICE ACTIVITY DETECTION
Publication number: 20140278383Abstract: Systems and methods are described to create a desired voice activity detection signal. A main acoustic signal and a plurality of reference acoustic signals are compressed. The compressed main acoustic signal is normalized by the plurality of compressed reference acoustic signals to create a plurality of normalized compressed main acoustic signals. The plurality of normalized compressed main acoustic signals is processed with a plurality of single channel normalized voice threshold comparators to form a plurality of normalized desired voice activity detection signals. One of the plurality of normalized desired voice activity detection signals is selected from the plurality of normalized desired voice activity detection signals to output as the desired voice activity detection signal.Type: ApplicationFiled: March 12, 2014Publication date: September 18, 2014Applicant: KOPIN CORPORATIONInventor: Dashen Fan -
Publication number: 20140278384Abstract: Systems and methods are described to automatically balance acoustic channel sensitivity. A long-term power level of a main acoustic signal is calculated to obtain an averaged main acoustic signal. Segments of the main acoustic signal are excluded from the averaged main acoustic signal using a desired voice activity detection signal. A long-term power level of a reference acoustic signal is calculated to obtain an averaged reference acoustic signal. Segments of the reference acoustic signal are excluded from the averaged reference acoustic signal using a desired voice activity detection signal. An amplitude correction signal is created using the averaged main acoustic signal and the averaged reference acoustic signal.Type: ApplicationFiled: March 12, 2014Publication date: September 18, 2014Applicant: KOPIN CORPORATIONInventor: Dashen Fan
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Patent number: 8818798Abstract: The invention relates to a method for determining a quality indicator representing a perceived quality of an output signal of an audio system with respect to a reference signal. The reference signal and the output signal are processed and compared. The processing includes dividing the reference signal and the output signal into mutually corresponding time frames, and includes scaling the intensity of the reference signal towards a fixed intensity level, and then performing measurements on time frames within the scaled reference signal for determining reference signal time frame characteristics. Further on, the loudness of the output signal is scaled towards a fixed loudness level in the perceptual loudness domain. Finally, the loudness of the reference signal is scaled from a loudness level corresponding to the output signal related intensity level towards a loudness level related to the loudness level of the scaled output signal in the perceptual loudness domain.Type: GrantFiled: August 9, 2010Date of Patent: August 26, 2014Assignees: Koninklijke KPN N.V., Nederlandse Organisatie voor Toegepast-Natuurwetenschappelijk Onderzoek TNOInventors: John Gerard Beerends, Jeroen van Vugt
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Patent number: 8805679Abstract: Provided are, among other things, systems, methods and techniques for detecting whether a transient exists within an audio signal. According to one representative embodiment, a segment of a digital audio signal is divided into blocks, and a norm value is calculated for each of a number of the blocks, resulting in a set of norm values for such blocks, each such norm value representing a measure of signal strength within a corresponding block. A maximum norm value is then identified across such blocks, and a test criterion is applied to the norm values. If the test criterion is not satisfied, a first signal indicating that the segment does not include any transient is output, and if the test criterion is satisfied, a second signal indicating that the segment includes a transient is output. According to this embodiment, the test criterion involves a comparison of the maximum norm value to a different second maximum norm value, subject to a specified constraint, within the segment.Type: GrantFiled: December 12, 2013Date of Patent: August 12, 2014Assignee: Digital Rise Technology Co., Ltd.Inventor: Yuli You
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Patent number: 8775163Abstract: A facility for conducting a real-time conversation in which the selected one of a number of participants utilizes a silent mode is described. Remark spoken by participants other than the selected one are transformed into text and displayed for the selected participant. Remarks entered textually by the selected participant are transformed into speech and played audibly for participants other than the selected one.Type: GrantFiled: March 15, 2013Date of Patent: July 8, 2014Inventors: Tony Bristol, Marvin Ingelman
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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
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Patent number: 8751225Abstract: Provided is an apparatus and method for encoding a voice and audio signal by expanding a modified discrete cosine transform (MDCT) based CODEC to a wideband and a super-wideband in a communication system. The apparatus for encoding a signal in a communication system, includes a converter configured to convert a time domain signal corresponding to a service to be provided to users to a frequency domain signal, a quantization and normalization unit configured to calculate and quantize gain of each subband in the converted frequency domain signal and normalize a frequency coefficient of the each subband, a search unit configured to search patch information of each subband in the converted frequency domain signal using the normalized frequency coefficient, and a packetizer configured to packetize the quantized gain and the searched patch information and encode gain information of each subband in the frequency domain signal.Type: GrantFiled: May 12, 2011Date of Patent: June 10, 2014Assignee: Electronics and Telecommunications Research InstituteInventors: Mi-Suk Lee, Hong-Kook Kim, Young-Han Lee
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Patent number: 8725518Abstract: A system for providing automatic quality management regarding a level of conformity to a specific accent, including, a recording system, a statistical model database with statistical models representing speech data of different levels of conformity to a specific accent, a speech analysis system, a quality management system. Wherein the recording system is adapted to record one or more samples of a speakers speech and provide it to the speech analysis system for analysis, and wherein the speech analysis system is adapted to provide a score of the speakers speech samples to the quality management system by analyzing the recorded speech samples relative to the statistical models in the statistical model database.Type: GrantFiled: April 25, 2006Date of Patent: May 13, 2014Assignee: Nice Systems Ltd.Inventors: Moshe Waserblat, Barak Eilam
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Patent number: 8700403Abstract: A method of statistical modeling is provided which includes constructing a statistical model and incorporating Gaussian priors during feature selection and during parameter optimization for the construction of the statistical model.Type: GrantFiled: November 3, 2005Date of Patent: April 15, 2014Assignee: Robert Bosch GmbHInventors: Fuliang Weng, Lin Zhao
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Patent number: 8655663Abstract: An audio signal interpolation device is presented, including an input unit for receiving an input audio signal, a phase splitting unit for splitting the input audio signal, a high range interpolation unit for interpolating a high range component into the signal, a phase combining unit for combining an in-phase component signal with a differential phase component, a high-pass filter for high-pass filtering the audio signal from by the phase combining unit, a delay unit for producing a delayed audio signal, and an addition processing unit for adding the delayed audio signal to the audio signal output from the high-pass filter.Type: GrantFiled: September 29, 2008Date of Patent: February 18, 2014Assignee: D&M Holdings, Inc.Inventors: Masaki Matsuoka, Shigeki Namiki
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Patent number: 8655652Abstract: An apparatus for encoding an information signal having discrete values includes a quantizer having a quantizer border, wherein the quantizer is adapted so that a discrete value above the quantization border is quantized to a quantization index, which is different from a quantization index obtained by quantizing a discrete value below the quantization border, a controller for modifying the quantization border, wherein the quantizer having a first quantization border setting is adapted to generate a first set of quantization indices for the discrete values, and wherein the quantizer having a second modified quantization border setting is adapted to generate a second set of quantization indices, and an output interface for outputting an encoded information signal which is either based on the first set of quantization indices or the second set of quantization indices dependent on a decision function.Type: GrantFiled: September 25, 2007Date of Patent: February 18, 2014Assignee: Dolby International ABInventor: Michael Schug
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Patent number: 8639517Abstract: Disclosed are systems, methods and computer-readable media for controlling a computing device to provide contextual responses to user inputs. The method comprises receiving a user input, generating a set of features characterizing an association between the user input and a conversation context based on at least a semantic and syntactic analysis of user inputs and system responses, determining with a data-driven machine learning approach whether the user input begins a new topic or is associated with a previous conversation context and if the received question is associated with the existing topic, then generating a response to the user input using information associated with the user input and any previous user input associated with the existing topic, based on a normalization of the length of the user input.Type: GrantFiled: June 15, 2012Date of Patent: January 28, 2014Assignee: AT&T Intellectual Property II, L.P.Inventors: Giuseppe Di Fabbrizio, Junlan Feng
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Patent number: 8634578Abstract: A multiband dynamics compressor implements a solution for minimizing unwanted changes to the long-term frequency response. The solution essentially proposes undoing the multiband compression in a controlled manner using much slower smoothing times. In this regard, the compensation provided acts more like an equalizer than a compressor. What is applied is a very slowly time-varying, frequency-dependent post-gain (make-up gain) that attempts to restore the smoothed long-term level of each compressor band.Type: GrantFiled: June 23, 2010Date of Patent: January 21, 2014Assignee: STMicroelectronics, Inc.Inventor: Earl C. Vickers
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Patent number: 8626499Abstract: A method or system for enabling client devices connected to a network to subscribe to a one or more multimedia signals available in a selection of bandwidths, based on an analysis of capabilities of the client, the quality of the connection, and the real-time changes in the ability of the client to process the signal.Type: GrantFiled: July 20, 2010Date of Patent: January 7, 2014Assignee: ViVu, Inc.Inventors: SivaKiran Venkata Yellamraju, Simha Sundeep Reddy Katasani
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Patent number: 8600764Abstract: Disclosed herein is a quantization method and apparatus of an audio encoder. The quantization method comprises calculating an absolute value of a maximum frequency spectrum of a first frame, externally received, by analyzing frequency spectrum data of the first frame, setting an initial value of a common scale factor to be used to quantize the first frame based on the absolute value of the maximum frequency spectrum of the first frame and an absolute value of a maximum frequency spectrum of a second frame, which has previously been calculated, and quantizing the frequency spectrum data of the first frame based on the set initial value of the common scale factor. Accordingly, before quantization is performed, an initial value of a common scale factor which is almost close to a value of an actual common scale factor can be previously set.Type: GrantFiled: March 3, 2010Date of Patent: December 3, 2013Assignee: Core Logic Inc.Inventor: Jae Mi Bahn
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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
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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
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Patent number: 8566086Abstract: A method and system for enhancing the frequency response of speech signals are provided. An average speech spectral shape estimate is calculated over time based on the input speech signal. The average speech spectral shape estimate may be calculated in the frequency domain using a first order IIR filtering or “leaky integrators.” Thus, the average speech spectral shape estimate adapts over time to changes in the acoustic characteristics of the voice path or any changes in the electrical audio path that may affect the frequency response of the system. A spectral correction factor may be determined by comparing the average speech spectral shape estimate to a desired target spectral shape. The spectral correction factor may be added (in units of dB) to the spectrum of the input speech signal in order to enhance or adjust the spectrum of the input speech signal toward the desired spectral shape, and an enhanced speech signal re-synthesized from the corrected spectrum.Type: GrantFiled: June 28, 2005Date of Patent: October 22, 2013Assignee: QNX Software Systems LimitedInventors: David Giesbrecht, Phillip Hetherington
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Patent number: 8560330Abstract: In accordance with an embodiment, A method of encoding an audio bitstream at an encoder includes encoding an original low band signal at the encoder by using a closed loop analysis-by-synthesis approach to obtain a coded low band signal, encoding an original high band signal at the encoder by using an open loop energy matching approach to obtain coded high band energy envelopes, comparing an energy of the coded low band signal with an energy of a corresponding original low band signal for a subframe, and generating an indication flag that indicates whether an energy envelope perceptual correction is needed for the subframe based on comparing the energy.Type: GrantFiled: July 19, 2011Date of Patent: October 15, 2013Assignee: Futurewei Technologies, Inc.Inventor: Yang Gao
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Publication number: 20130226571Abstract: A method for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder receives encoded frames of compressed speech information transmitted from an encoder. The method determines whether an encoded frame has been lost, corrupted in transmission, or erased, synthesizes properly received frames, and decides on an overlap-add window to use in combining a portion of the synthesized speech signal with a subsequent speech signal resulting from a received and decoded packet, where the size of the overlap-add window is based on the unavailability of packets. If it is determined that an encoded frame has been lost, corrupted in transmission, or erased, the method performed an overlap-add operation on the portion of the synthesized speech signal and the subsequent speech signal, using the decided-on overlap-add window.Type: ApplicationFiled: April 15, 2013Publication date: August 29, 2013Applicant: AT&T Intellectual Property II, L.P.Inventor: AT&T Intellectual Property II, L.P.
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Patent number: 8521522Abstract: There is provided an audio coding device which appropriately sets the quantization bit number by a small calculation amount in each stage when coding an input audio signal by performing multi-stage normalization/quantization. A quantization information calculation section determines total quantization information idwl0, based on normalization information idsf, and allocates the total quantization information idwl0 for quantization information idwl1 and quantization information idwl2. At this time, the quantization information calculation section limits the quantization information idwl1 by a limiter lim1, and allocates the total quantization information idwl0 for quantization information idwl1. If the quantization information idwl1 exceeds the limiter lim1, the excess is allocated for the quantization information idwl2. A first normalization section and a first quantization section normalizes and quantizes a frequency spectrum mdspec1 in the first stage.Type: GrantFiled: May 5, 2006Date of Patent: August 27, 2013Assignee: Sony CorporationInventors: Yuuki Matsumura, Shiro Suzuki, Keisuke Toyama, Mitsuyuki Hatanaka, Yuhki Mitsufuji
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Patent number: 8510104Abstract: A system and method are provided to authenticate a voice in a frequency domain. A voice in the time domain is transformed to a signal in the frequency domain. The first harmonic is set to a predetermined frequency and the other harmonic components are equalized. Similarly, the amplitude of the first harmonic is set to a predetermined amplitude, and the harmonic components are also equalized. The voice signal is then filtered. The amplitudes of each of the harmonic components are then digitized into bits to form at least part of a voice ID. In another system and method, a voice is authenticated in a time domain. The initial rise time, initial fall time, second rise time, second fall time and final oscillation time are digitized into bits to form at least part of a voice ID. The voice IDs are used to authenticate a user's voice.Type: GrantFiled: September 14, 2012Date of Patent: August 13, 2013Assignee: Research In Motion LimitedInventor: Sasan Adibi
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Patent number: 8484022Abstract: A method and system for adaptive auto-encoders is disclosed. An input audio training signal may be transformed into a sequence of feature vectors, each bearing quantitative measures of acoustic properties of the input audio training signal. An auto-encoder may process the feature vectors to generate an encoded form of the quantitative measures, and a recovered form of the quantitative measures based on an inverse operation by the auto-encoder on the encoded form of the quantitative measures. A duplicate copy of the sequence of feature vectors may be normalized to form a normalized signal in which supra-phonetic acoustic properties are reduced in comparison with phonetic acoustic properties of the input audio training signal. The auto-encoder may then be trained to compensate for supra-phonetic features by reducing the magnitude of an error signal corresponding to a difference between the normalized signal and the recovered form of the quantitative measures.Type: GrantFiled: July 27, 2012Date of Patent: July 9, 2013Assignee: Google Inc.Inventor: Vincent Vanhoucke
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Patent number: 8484267Abstract: Weight normalization in hardware or software without a division operator is described, using only right bit shift, addition and subtraction operations. A right bit shift is performed on an expected sum to effectively divide the expected sum by two to provide a first updated value for the expected sum. An iteration is performed which includes: incrementing with a first adder a first variable by the first updated value of the expected sum to provide an updated value for the first variable; subtracting with a first subtractor a second weight from a first weight to provide a first updated value for the first weight; and performing a left bit shift on the second weight to effectively multiply the second weight by two to provide a first updated value for the second weight.Type: GrantFiled: November 19, 2009Date of Patent: July 9, 2013Assignee: Xilinx, Inc.Inventor: Gabor Szedo
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Patent number: 8478586Abstract: A coded code string from an input terminal 110 is demultiplexed by a demultiplexer circuit 101, normalization coefficient information in the code string is sent to a normalization coefficient information increasing/decreasing circuit 102, addition or subtraction of a positive value is performed, and level adjustment of a signal is performed. A normalization coefficient information cutoff amount calculating circuit 103 calculates the cutoff amount for a case where the subtraction amount of normalization coefficient information is larger than normalization coefficient information and normalization coefficient information after subtraction is cut off at the minimum possible value. A gain control function generation information modifying circuit 104 modifies gain control function generation information according to the cutoff amount.Type: GrantFiled: June 26, 2008Date of Patent: July 2, 2013Assignee: Sony CorporationInventor: Hiroyuki Honma
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Patent number: 8478599Abstract: An embodiment of the present invention is a method of presenting a media work which includes: detecting media work content properties in a portion of the media work; associating a presentation rate of the portion with the detected media work content properties; and presenting the portion at the presentation rate; wherein the media work content properties include one or more of: (a) indicia of a number of syllables in utterances; (b) indicia of a number of letters in a word; (c) indicia of the complexity of grammatical structures in portions of the media work; (d) indicia of arrival rate of newly presented objects; (e) indicia of temporal proximity of between events in portions of the media work or (f) indicia of number of phonemes per unit of time in portions of the media work.Type: GrantFiled: May 18, 2009Date of Patent: July 2, 2013Assignee: Enounce, Inc.Inventor: Donald J. Hejna, Jr.
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Patent number: 8471742Abstract: A device for continuous time quantization of an input signal, in order to supply a continuous time output signal that is quantized as two bits, the device including: an electronic circuit, designed to supply a first bit of the output signal called the sign bit which at any time takes a first value when the input signal is positive and a second value when the input signal is negative, and an envelope analysis circuit designed to supply a second bit of the output signal called the envelope variation bit which at any time takes a first value, called high value, when an envelope signal of the input signal is increasing, and a second value, called low value, when the envelope signal is decreasing.Type: GrantFiled: April 20, 2011Date of Patent: June 25, 2013Assignee: Commissariat a l'Energie Atomique et aux Energies AlternativesInventor: David Lachartre
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Patent number: 8463614Abstract: An audio encoding method and a corresponding decoding method are provided. Accordingly, the pre-echo effect of the audio transient signal is eliminated and the distortion of the transient signal is mitigated. The technical solution includes performing time-domain processing on an input audio transient signal; dividing sampling points x1,x2, . . . , xN of an input frame into L segments; calculating an energy Ei for each segment; calculating an average energy E0 for each segment of the input frame; calculating a multiplying parameter ?i corresponding to each segment by virtue of ?i=r(bitrate)*E0/Ei; multiplying the sampling points of all the segments of the input frame by corresponding multiplying parameter ?i, obtaining the processed sampling points x1?,x2?, . . . , xN?; and sending the multiplying parameter ?i to a code stream for transportation; performing time-frequency transformation and coding on the processed sampling points x1?,x2?, . . . , xN? and outputting to the code stream.Type: GrantFiled: November 10, 2009Date of Patent: June 11, 2013Assignee: Spreadtrum Communications (Shanghai) Co., Ltd.Inventors: Benhao Zhang, Heyun Huang, Tan Li, Fuhuei Lin
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Patent number: 8457954Abstract: According to one embodiment, there is provided a sound quality control apparatus, including: a characteristic parameter extractor; a speech score calculator; a music score calculator; a power value acquisition module; a first storage configured to store speech scores and music scores; a second storage configured to store power values; a power-based score corrector configured to correct a current music score or a current speech score based on a first comparison result between a current power value and past power values, a second comparison result between the current music score and past music scores and a third comparison result between the current speech score and past speech scores; and a sound quality controller configured to perform a sound quality control by using at least one of the speech score and the music score corrected by the power-based score corrector.Type: GrantFiled: April 28, 2011Date of Patent: June 4, 2013Assignee: Kabushiki Kaisha ToshibaInventors: Hirokazu Takeuchi, Hiroshi Yonekubo
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Patent number: 8447595Abstract: A method for performing a call between a near-end user and a far-end user, which includes the following operations performed during the call by the near-end user's communications device. Automatic gain control (AGC) is performed to update a gain applied to an uplink speech signal. A frame is detected in a downlink signal that contains speech; in response, the updating of the gain is frozen. Other embodiments are also described and claimed.Type: GrantFiled: June 3, 2010Date of Patent: May 21, 2013Assignee: Apple Inc.Inventor: Shaohai Chen
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Patent number: 8442819Abstract: Methods and apparatus are disclosed for controlling a buffer in a communication system, such as a digital audio broadcasting (DAB) communication system. A more consistent perceptual quality over time provides for a more pleasing auditory experience to a listener. The disclosed bit allocation process determines, for each frame, a distortion d[k] at which the frame is to be encoded. The distortion d[k] is determined to minimize (i) the probability for a buffer overflow, and (ii) the variation of perceived distortion over time. A buffer level is controlled by partitioning a signal into a sequence of successive frames; estimating a distortion rate for a number of frames; and selecting a distortion such that the variance of the buffer level is bounded by a specified value.Type: GrantFiled: April 13, 2006Date of Patent: May 14, 2013Assignee: Agere Systems LLCInventor: Christof Faller
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Patent number: 8423357Abstract: Embodiments of the invention provide a communication device and methods for generating enhanced audio signals. An audio signal comprising a speech signal and a noise signals is acquired at the communication device. A noise processor of the communication device detects a pitch estimation of the audio signal. Thereafter, the audio signal is processed based on the pitch estimation and processing parameters of the audio signals to remove noise signals and generate an enhanced audio signal.Type: GrantFiled: June 16, 2011Date of Patent: April 16, 2013Inventors: Sandeep Kulakcherla, Alon Konchitsky, Alberto D Berstein
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Patent number: 8401844Abstract: Disclosed is a gain control system in which speech model constituted from a sound pressure and a feature is stored in a speech model storage unit for each of a plurality of phonemes or for each of clusters into which a speech is divided. When an input signal is given, a feature conversion unit calculates a feature and a sound pressure of the input signal. A sound pressure comparison unit determines a sound pressure ratio between the input signal and each of speech models. A distance calculation unit calculates a distance between the feature of the input signal and the feature of each of the speech models. A gain calculation unit calculates a gain value from the sound pressure ratio and information on the distance. A sound pressure compensation unit thereby compensates for the sound pressure of the input signal.Type: GrantFiled: January 16, 2007Date of Patent: March 19, 2013Assignee: NEC CorporationInventors: Takayuki Arakawa, Masanori Tsujikawa
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Patent number: 8391373Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.Type: GrantFiled: March 20, 2009Date of Patent: March 5, 2013Assignee: France TelecomInventors: David Virette, Pierrick Philippe, Balazs Kovesi
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Patent number: 8379880Abstract: An example of a method of correcting an audio level of a stored program asset comprises retrieving a stored program asset having audio encoded at a first loudness setting. Dialog of the audio of the asset is identified, a loudness of the dialog is determined and the determined loudness is compared to the first loudness setting. The asset is re-encoded at a second loudness setting corresponding to the determined loudness, if the first loudness setting and the second loudness are different by more than a predetermined amount. The determined loudness is preferably a DIALNORM of the dialog. The asset may be stored with the re-encoded loudness setting. The method may be applied to programs as they are being received from a source, as well. Aspects of the method may also be applied to programs to be provided by a source. Systems are also disclosed.Type: GrantFiled: June 2, 2008Date of Patent: February 19, 2013Assignee: Time Warner Cable Inc.Inventor: Steven E. Riedl
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Patent number: 8374853Abstract: A system for coding a hierarchical audio signal, comprising, at least, a core layer using parametric coding by analysis by synthesis in a first frequency band, a band extension layer for widening said first frequency band into a second frequency band, or wideband. The system also comprises a wideband audio coding quality enhancement layer based on transform coding using a spectral parameter obtained from said band extension layer. Application to transmitting speech and/or audio signals over packet networks.Type: GrantFiled: July 7, 2006Date of Patent: February 12, 2013Assignee: France TelecomInventors: Stéphane Ragot, David Virette
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Patent number: 8364477Abstract: A method (400, 500) and apparatus (220) seeks to improve the intelligibility of speech emitted into a noisy environment. Formants are identified (426) and perceptual frequency scale band is selected (502) that includes at least one of the identified formants. The SNR in each band is compared (504) to a threshold and, if the SNR for that band is less than the threshold, the method increases a formant enhancement gain for that band. A set of high pass filter gains (338) is combined (516) with the formant enhancement gains yielding combined gains that are then clipped (518), scaled (520) according to a total SNR, normalized (526), smoothed across time (530) and frequency (532), and used to reconstruct (532, 534) an audio signal.Type: GrantFiled: August 30, 2012Date of Patent: January 29, 2013Assignee: Motorola Mobility LLCInventors: Jianming J Song, John C Johnson
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Patent number: 8355908Abstract: In frequency signals obtained by converting input audio signals from time-domain signals to frequency-domain signals, a level control value setting unit 5 establishes a level control value for reducing the levels of spectrums at a noise-components level. A level control value smoothing unit 6 carries out a smoothing process of smoothing the level control value established by the level control value setting unit 5 temporally. A spectral adjustment unit 8 multiplies the level control value after the smoothing process by the frequency signals, performing a level control.Type: GrantFiled: March 19, 2009Date of Patent: January 15, 2013Assignee: JVC Kenwood CorporationInventors: Takaaki Yamabe, Masaya Konishi
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Patent number: 8352256Abstract: An audio input signal is filtered using an adaptive filter to generate a prediction output signal with reduced noise, wherein the filter is implemented using a plurality of coefficients to generate a plurality of prediction errors and to generate an error from the plurality of prediction errors, wherein the absolute values of the coefficients are continuously reduced by a plurality of reduction parameters.Type: GrantFiled: September 30, 2010Date of Patent: January 8, 2013Assignee: Entropic Communications, Inc.Inventor: Joern Fischer
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Patent number: 8353058Abstract: A computer-implemented method for detecting rootkits is disclosed. The computer-implemented method may include sending periodic security communications from a privileged-processor-mode region of a computing device. The computer-implemented method may also include identifying at least one of the periodic security communications. The computer-implemented method may further include determining, based on the periodic security communications, whether the privileged-processor-mode region of the computing device has been compromised. Various other methods, systems, and computer-readable media are also disclosed.Type: GrantFiled: March 24, 2009Date of Patent: January 8, 2013Assignee: Symantec CorporationInventors: Bruce McCorkendale, Sourabh Satish, William E. Sobel
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Patent number: 8346542Abstract: An audio signal band expanding apparatus (100a) includes a harmonic generator (3) that receives an input audio signal having a predetermined band and generates, based on the input audio signal, harmonic signals, and an adder (2) that adds the harmonic signals generated by the harmonic generator (3) to the input audio signal. The harmonic generator (3) simulates the input-output characteristics of a predetermined amplifier or that of a device to generate the harmonic signals from the input audio signal.Type: GrantFiled: February 16, 2012Date of Patent: January 1, 2013Assignee: Panasonic CorporationInventor: Kazuya Iwata
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Publication number: 20120323570Abstract: Described herein are methods, systems, apparatuses and products for reconstruction of a smooth speech signal from a stuttered speech signal. One aspect provides for accessing a stored speech signal having stuttering; identifying at least one stuttered region in the stored speech signal; modifying the at least one stuttered region in the stored speech signal; and responsive to modifying the at least one stuttered region, reconstructing a smooth speech signal corresponding to the stored speech signal. Other embodiments are disclosed.Type: ApplicationFiled: August 28, 2012Publication date: December 20, 2012Applicant: INTERNATIONAL BUSINESS MACHINES CORPORATIONInventors: Om Dadaji Deshmukh, Suraj Satishkumar Sheth, Ashish Verma
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Patent number: 8321209Abstract: A system and method are provided to authenticate a voice in a frequency domain. A voice in the time domain is transformed to a signal in the frequency domain. The first harmonic is set to a predetermined frequency and the other harmonic components are equalized. Similarly, the amplitude of the first harmonic is set to a predetermined amplitude, and the harmonic components are also equalized. The voice signal is then filtered. The amplitudes of each of the harmonic components are then digitized into bits to form at least part of a voice ID. In another system and method, a voice is authenticated in a time domain. The initial rise time, initial fall time, second rise time, second fall time and final oscillation time are digitized into bits to form at least part of a voice ID. The voice IDs are used to authenticate a user's voice.Type: GrantFiled: November 10, 2009Date of Patent: November 27, 2012Assignee: Research In Motion LimitedInventor: Sasan Adibi
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Patent number: 8315853Abstract: A post-filtering apparatus and method for speech enhancement in a modified discrete cosine transform (MDCT) domain are disclosed. In the apparatus and method, previous and current MDCT coefficients are used for obtaining a speech spectrum coefficient similar to a real speech spectrum, and a convex function is used for transforming the speech spectrum coefficient and obtaining a post-filter coefficient so that difference can increase in the case where the speech spectrum coefficient is small but decrease in the case where the coefficient is large. Then, the post-filter coefficient is applied to the MDCT coefficient. With this configuration, both the current and previous MDCT values are used, so that it is possible to obtain a spectrum coefficient similar to the real speech spectrum and to obtain a more accurate filter coefficient. Further, the coefficient is adaptively transformed through the convex function, thereby enhancing speech quality.Type: GrantFiled: June 5, 2008Date of Patent: November 20, 2012Assignee: Electronics and Telecommunications Research InstituteInventors: Hyun-woo Kim, Jong-mo Sung, Mi-suk Lee, Do-young Kim, Byung-sun Lee
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Patent number: 8315859Abstract: A filter apparatus for filtering a time domain input signal to obtain a time domain output signal, which is a representation of the time domain input signal filtered using a filter characteristic having an non-uniform amplitude/frequency characteristic, comprises a complex analysis filter bank for generating a plurality of complex subband signals from the time domain input signals, a plurality of intermediate filters, wherein at least one of the intermediate filters of the plurality of the intermediate filters has a non-uniform amplitude/frequency characteristic, wherein the plurality of intermediate filters have a shorter impulse response compared to an impulse response of a filter having the filter characteristic, and wherein the non-uniform amplitude/frequency characteristics of the plurality of intermediate filters together represent the non-uniform filter characteristic, and a complex synthesis filter bank for synthesizing the output of the intermediate filters to obtain the time domain output signal.Type: GrantFiled: March 17, 2010Date of Patent: November 20, 2012Assignee: Dolby International ABInventor: Lars Villemoes
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Patent number: 8280730Abstract: A method (400, 600, 700) and apparatus (220) for enhancing the intelligibility of speech emitted into a noisy environment. After filtering (408) ambient noise with a filter (304) that simulates the physical blocking of noise by a at least a part of a voice communication device (102) a frequency dependent SNR of received voice audio relative to ambient noise is computed (424) on a perceptual (e.g. Bark) frequency scale. Formants are identified (426, 600, 700) and the SNR in bands including certain formants are modified (508, 510) with formant enhancement gain factors in order to improve intelligibility. A set of high pass filter gains (338) is combined (516) with the formant enhancement gains factors yielding combined gains which are clipped (518), scaled (520) according to a total SNR, normalized (526), smoothed across time (530) and frequency (532) and used to reconstruct (532, 534) an audio signal.Type: GrantFiled: May 25, 2005Date of Patent: October 2, 2012Assignee: Motorola Mobility LLCInventors: Jianming J. Song, John C. Johnson
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Publication number: 20120232889Abstract: A method for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder receives encoded frames of compressed speech information transmitted from an encoder. The method determines whether an encoded frame has been lost, corrupted in transmission, or erased, synthesizes properly received frames, and decides on an overlap-add window to use in combining a portion of the synthesized speech signal with a subsequent speech signal resulting from a received and decoded packet, where the size of the overlap-add window is based on the unavailability of packets. If it is determined that an encoded frame has been lost, corrupted in transmission, or erased, the method performed an overlap-add operation on the portion of the synthesized speech signal and the subsequent speech signal, using the decided-on overlap-add window.Type: ApplicationFiled: May 21, 2012Publication date: September 13, 2012Applicant: AT&T Corp.Inventor: David A. Kapilow
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Patent number: 8265929Abstract: Provides is an embedded code-excited linear prediction speech coding/decoding apparatus and method that can deal with the capacity change of speech transmission channel by modeling an error signal not coded at a core speech coder based on a transmission rate in a multiple pulse search mode or gain compensation mode and then transmitting it in an optimum mode. The apparatus includes a core speech coding unit for coding an input speech signal with spectral envelop and an excitation signal, a transmission rate determination unit for allocating the number of bits additionally allowed depending on a capacity of a transmission channel, and an embedded excitation signal coding unit for coding a residual excitation signal that is not coded in the core speech coding unit based on the number of additionally allowed bits using one of a multiple pulse excitation coding mode and a gain compensation mode.Type: GrantFiled: December 7, 2005Date of Patent: September 11, 2012Assignee: Electronics and Telecommunications Research InstituteInventors: Mi-Suk Lee, Do-Young Kim, JongMo Sung, Hyun-Woo Kim
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Patent number: 8255215Abstract: It is an object of the present invention to provide a method and apparatus for locating a keyword of a speech and a speech recognition system. The method includes the steps of: by extracting feature parameters from frames constituting the recognition target speech, forming a feature parameter vector sequence that represents the recognition target speech; by normalizing of the feature parameter vector sequence with use of a codebook containing a plurality of codebook vectors, obtaining a feature trace of the recognition target speech in a vector space; and specifying the position of a keyword by matching prestored keyword template traces with the feature trace. According to the present invention, a keyword template trace and a feature space trace of a recognition target speech are drawn in accordance with an identical codebook. This causes resampling to be unnecessary in performing linear movement matching of speech wave frames having similar phonological feature structures.Type: GrantFiled: September 27, 2007Date of Patent: August 28, 2012Assignee: Sharp Kabushiki KaishaInventors: Fengqin Li, Yadong Wu, Qinqtao Yang, Chen Chen