Using Predictive Techniques; Codecs Based On Source-filter Modelization (epo) Patents (Class 704/E19.023)
  • Publication number: 20140129214
    Abstract: A method (1100) and apparatus (100) generate a candidate code-vector to code an information signal. The method can include producing (1110) a weighted target vector from an input signal. The method can include processing (1120) the weighted target vector through an inverse weighting function to create a residual domain target vector. The method can include performing (1130) a first search process on the residual domain target vector to obtain an initial fixed codebook code-vector. The method can include performing (1140) a second search process over a subset of possible codebook code-vectors for a low weighted-domain error to produce a final fixed codebook code-vector. The subset of possible codebook code-vectors can be based on the initial fixed codebook code-vector. The method can include generating (1150) a codeword representative of the final fixed codebook code-vector. The codeword can be for use by a decoder to generate an approximation of the input signal.
    Type: Application
    Filed: November 2, 2012
    Publication date: May 8, 2014
    Applicant: Motorola Mobility LLC
    Inventors: James P. Ashley, Udar Mittal
  • Publication number: 20140088974
    Abstract: A method and apparatus provides for frame loss recovery following a loss of a frame in an audio codec. The lost frame is identified. Estimated linear predictive coefficients of a previous transform frame are generated based on a decoded audio of the previous transform frame. An estimated residual of the previous transform frame is generated based on the estimated linear predicative coefficients and the decoded audio. A pitch delay is determined from frame error recovery parameters received with the previous transform frame. An extended residual is generated based on the pitch delay and the estimated residual. A first synthesized signal is generated based on the extended residual and the linear predicative coefficients. A decoded audio output of at least the lost frame is generated based on the first synthesized signal. The frame error recovery parameters are generated by an encoder.
    Type: Application
    Filed: September 26, 2012
    Publication date: March 27, 2014
    Applicant: MOTOROLA MOBILITY LLC
    Inventors: Udar Mittal, James P. Ashley
  • Publication number: 20130268266
    Abstract: A method (300) and apparatus (100) generate a candidate code-vector to code an information signal. The method can include producing (310) a target vector from a received input signal. The method can include constructing (320) a plurality of inverse weighting functions based on the target vector. The method can include evaluating (330) an error value associated with each of the plurality of inverse weighting functions to produce a fixed codebook code-vector. The method can include generating (340) a codeword representative of the fixed codebook code-vector, where the codeword can be used by a decoder to generate an approximation of the input signal.
    Type: Application
    Filed: April 4, 2012
    Publication date: October 10, 2013
    Applicant: Motorola Mobility, Inc.
    Inventors: James P. Ashley, Udar Mittal
  • Patent number: 8515743
    Abstract: A method and apparatus for searching fixed codebook are provided. The method includes: obtaining a basic codebook which comprises position information of N pulses on M tracks, wherein N and M are positive integers; choosing n pulses as search pulses, wherein the n pulses are parts of the N pulses and n is a positive integer smaller than N; and replacing position information of the n search pulses respectively with other position information on the tracks to obtain a searched codebook; executing the search process for K times, wherein K is a positive integer larger than or equal to 2, at least two or more search pulses are chosen in one of the K search processes , and the chosen search pulses vary in each of the K search processes; and obtaining an optimal codebook from the basic codebook and the searched codebook according to a preset criterion.
    Type: Grant
    Filed: June 4, 2009
    Date of Patent: August 20, 2013
    Assignee: Huawei Technologies Co., Ltd
    Inventors: Dejun Zhang, Lixiong Li
  • Publication number: 20130179159
    Abstract: A method for detecting overflow on an electronic device is described. The method includes determining a linear predictive coding synthesis filter gain. The method further includes determining whether overflow is detected based on the linear predictive coding synthesis filter gain and a fixed codebook gain. The method further includes determining a scaling factor if overflow is detected.
    Type: Application
    Filed: November 1, 2012
    Publication date: July 11, 2013
    Applicant: QUALCOMM Incorporated
    Inventor: QUALCOMM Incorporated
  • Publication number: 20130096913
    Abstract: There is provided an apparatus and method for encoding a speech signal. The encoding comprises: receiving a plurality of current samples of the speech signals; extrapolating a plurality of look-ahead samples from the current samples; and performing linear prediction analysis using the current samples and the extrapolated look-ahead samples.
    Type: Application
    Filed: November 30, 2011
    Publication date: April 18, 2013
    Applicant: TELEFONAKTIEBOLAGET L M ERICSSION (publ)
    Inventors: Stefan BRUHN, Jingming Kuang, Jing Wang, Chunling Zhang, Shenghui Zhao
  • Publication number: 20130054230
    Abstract: Methods, systems, and non-transitory computer readable media for estimating speech energy of an encoded bit stream based on coding parameters extracted from the partially-decoded bit stream are disclosed. According to one aspect, a method for estimating speech energy of an encoded bit stream based on coding parameters extracted from the partially-decoded bit stream includes receiving a CELP-encoded bit stream, partially decoding the bit stream, and estimating the speech energy of the bit stream based a set of four or fewer CELP parameters extracted from the partially decoded bit stream.
    Type: Application
    Filed: August 22, 2011
    Publication date: February 28, 2013
    Inventors: Emmanuel Rossignol Thepie Fapi, Eric Poulin, Jean Pierre Doyon
  • Publication number: 20130030800
    Abstract: Systems and methods for adaptively processing speech to improve voice intelligibility are described. These systems and methods can adaptively identify and track formant locations, thereby enabling formants to be emphasized as they change. As a result, these systems and methods can improve near-end intelligibility, even in noisy environments. The systems and methods can be implemented in Voice-over IP (VoIP) applications, telephone and/or video conference applications (including on cellular phones, smart phones, and the like), laptop and tablet communications, and the like. The systems and methods can also enhance non-voiced speech, which can include speech generated without the vocal track, such as transient speech.
    Type: Application
    Filed: July 26, 2012
    Publication date: January 31, 2013
    Applicant: DTS, LLC
    Inventors: James Tracey, Daekyong Noh, Xing He
  • Publication number: 20130030799
    Abstract: An acoustic shock protection device includes a prediction gain estimator and an audio compressor. The prediction gain estimator is configured to analyze a plurality of linear prediction coefficients of an audio signal and determine a category of the audio signal. The audio compressor is coupled to the prediction gain estimator, and the audio compressor is configured to adjust a signal level of the audio signal according to the category of the audio signal.
    Type: Application
    Filed: July 25, 2012
    Publication date: January 31, 2013
    Applicant: VIA TELECOM, INC.
    Inventors: Meoung-Jin Lim, Sanghyun Chi
  • Publication number: 20130024191
    Abstract: An audio communication device comprises an input connectable to a narrowband audio signal source. The input can receive a narrowband audio signal having a first bandwidth. An extraction unit is connected to the input and arranged to extract a plurality of narrowband parameters from the narrowband audio signal. An extrapolation unit is connected to receive the plurality of narrowband parameters and arranged to generate a plurality of wideband parameters from the plurality of narrowband parameters. The extrapolation unit comprises one or more adaptive neuro-fuzzy inference system modules. The device further comprises a synthesis unit connected to receive the plurality of wideband parameters and arranged to generate, using the wideband parameters, a synthesized wideband audio signal having a second bandwidth wider than the first bandwidth.
    Type: Application
    Filed: April 12, 2010
    Publication date: January 24, 2013
    Applicant: FREESCALE Semiconductor, Inc.
    Inventors: Robert Krutsch, Radu D. Pralea
  • Patent number: 8346566
    Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.
    Type: Grant
    Filed: August 31, 2010
    Date of Patent: January 1, 2013
    Assignee: Dolby International AB
    Inventors: Kristofer Kjoerling, Lars Villemoes
  • Publication number: 20120303362
    Abstract: A method of noise-robust speech classification is disclosed. Classification parameters are input to a speech classifier from external components. Internal classification parameters are generated in the speech classifier from at least one of the input parameters. A Normalized Auto-correlation Coefficient Function threshold is set. A parameter analyzer is selected according to a signal environment. A speech mode classification is determined based on a noise estimate of multiple frames of input speech.
    Type: Application
    Filed: April 10, 2012
    Publication date: November 29, 2012
    Applicant: QUALCOMM Incorporated
    Inventors: Ethan Robert Duni, Vivek Rajendran
  • Publication number: 20120278069
    Abstract: A quantizing method is provided that includes quantizing an input signal by selecting one of a first quantization scheme not using an inter-frame prediction and a second quantization scheme using the inter-frame prediction, in consideration of one or more of a prediction mode, a predictive error and a transmission channel state.
    Type: Application
    Filed: April 23, 2012
    Publication date: November 1, 2012
    Applicant: SAMSUNG ELECTRONICS CO., LTD.
    Inventors: Ho-sang SUNG, Eun-mi OH
  • Publication number: 20120253797
    Abstract: In an embodiment, bitstream elements of sub-frames are encoded differentially to a global gain value so that a change of the global gain value results in an adjustment of an output level of the decoded representation of the audio content. Concurrently, the differential coding saves bits. Even further, the differential coding enables the lowering of the burden of globally adjusting the gain of an encoded bitstream. In another embodiment, a global gain control across CELP coded frames and transform coded frames is achieved by co-controlling the gain of the codebook excitation of the CELP codec, along with a level of the transform or inverse transform of the transform coded frames. In another embodiment, the gain value determination in CELP coding is performed in the weighted domain of the excitation signal.
    Type: Application
    Filed: April 18, 2012
    Publication date: October 4, 2012
    Inventors: Ralf Geiger, Guillaume Fuchs, Markus Multrus, Bernhard Grill
  • Publication number: 20120245930
    Abstract: According to the present invention, a linear prediction filter coefficient of a current frame is acquired from an input signal using linear prediction, a quantized spectrum candidate vector of the current frame, corresponding to the linear prediction filter coefficient of the current frame, is acquired on the basis of first best information, and the quantized spectrum candidate vector of the current frame and the quantized spectrum vector of the previous frame are interpolated. Accordingly, in contrast to conventional phased optimization techniques, optimum parameters which minimize quantization errors, can be obtained.
    Type: Application
    Filed: December 10, 2010
    Publication date: September 27, 2012
    Applicant: LG ELECTRONICS INC.
    Inventors: Hyejeong Jeon, Daehwan Kim, Gyuhyeok Jeong, Minki Lee, Honggoo Kang, Byungsuk Lee, Lagyoung Kim
  • Publication number: 20120239408
    Abstract: A method of processing an audio signal is disclosed.
    Type: Application
    Filed: September 17, 2010
    Publication date: September 20, 2012
    Applicants: LG ELECTRONICS INC., INDUSTRY-ACADEMIC COOPERATION FOUNDATION, YONSEI UNIVERSITY
    Inventors: Hyen-O Oh, Chang Heon Lee, Hong Goo Kang, Jeongook Song
  • Publication number: 20120232888
    Abstract: An apparatus and method for concealing frame erasure and a voice decoding apparatus and method using the same. The frame erasure concealment apparatus includes: a parameter extraction unit determining whether there is an erased frame in a voice packet, and extracting an excitement signal parameter and a line spectrum pair parameter of a previous good frame; and an erasure frame concealment unit, if there is an erased frame, restoring the excitement signal and line spectrum pair parameter of the erased frame by using a regression analysis from the excitement signal and line spectrum pair parameter of the previous good frame. According to the method and apparatus, by predicting and restoring the parameter of the erased frame through the regression analysis, the quality of the restored voice signal can be enhanced and the algorithm can be simplified.
    Type: Application
    Filed: May 22, 2012
    Publication date: September 13, 2012
    Inventors: Hosang Sung, Kangeun Lee, Seungho Choi
  • Publication number: 20120185241
    Abstract: An audio decoding apparatus comprises: a plurality of decoding units; a band replicating unit which processes a decoded signal obtained when a corresponding decoding unit decodes a coded signal, according to a scheme specified by transmitted information; and an information transmitting unit which transmits, to a signal processing unit, information identifying the corresponding decoding unit from among the plurality of decoding units.
    Type: Application
    Filed: March 28, 2012
    Publication date: July 19, 2012
    Applicant: Panasonic Corporation
    Inventors: Shuji MIYASAKA, Kosuke Nishio, Takeshi Norimatsu
  • Publication number: 20120173246
    Abstract: The present invention provides a new recursive FIR filter scheme which supports a variable order short-term predictor, and uses a pipeline stall based on the radix-2 algorithm and an autocorrelation processing time for reducing the complexity of MPEG-4 ALS hardware implementation.
    Type: Application
    Filed: December 29, 2011
    Publication date: July 5, 2012
    Applicant: KOREA ELECTRONICS TECHNOLOGY INSTITUTE
    Inventors: Byeong Ho CHOI, Dong Sun KIM, Je Woo KIM, Choong Sang CHO, Seung Yerl LEE, Sang Seol LEE
  • Publication number: 20120095758
    Abstract: A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.
    Type: Application
    Filed: September 28, 2011
    Publication date: April 19, 2012
    Applicant: MOTOROLA MOBILITY, INC.
    Inventors: Jonathan A. Gibbs, James P. Ashley, Udar Mittal
  • Publication number: 20120010879
    Abstract: A linear prediction coefficient of a signal represented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorrelation method. After the filter strength of the obtained linear prediction coefficient is adjusted, filtering may be performed in the frequency direction on the signal by using the adjusted coefficient, whereby the temporal envelope of the signal is transformed. This reduces the occurrence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a band extension technique in the frequency domain represented by SBR.
    Type: Application
    Filed: September 23, 2011
    Publication date: January 12, 2012
    Applicant: NTT DOCOMO, INC.
    Inventors: Kosuke Tsujino, Kei Kikuiri, Nobuhiko Naka
  • Publication number: 20110301947
    Abstract: Packets of real-time information are sent with a source rate greater than zero kilobits per second, and a time or path or combined time/path diversity rate initially being zero kilobits per second. This results in a quality of service QoS, optionally measured at the sender or the receiver. When the QoS is on an unacceptable side of a threshold of acceptability, the sender sends diversity packets at an increased rate. Increasing the diversity rate while either reducing or maintaining the overall transmission rate is new. CELP-based multiple-description data partitioning sends the base or important information plus a subset of fixed excitation in one packet and sends the base or important information plus the complementary subset of fixed excitation in another packet. Reconstruction produces acceptable quality when only one of the two packets is received and better quality when both packets are received. Reconstruction provides for single and multiple lost packets.
    Type: Application
    Filed: August 15, 2011
    Publication date: December 8, 2011
    Applicant: TEXAS INSTRUMENTS INCORPORATED
    Inventors: Krishnasamy Anandakumar, Vishu R. Viswanathan, Alan V. McCree
  • Publication number: 20110166854
    Abstract: A method and apparatus multiplies a past sample a time lag ? older than a current sample by a quantized multiplier ?? on a frame by frame basis, subtracts the multiplication result from the current sample, codes the subtraction result, and codes the time lag using a fixed-length coder if the multiplier ?? is smaller than 0.2 or if information about the previous frame is unavailable, or codes the time lag using a variable-length coder if ?? is not smaller than 0.2. A multiplier ? is coded by a multiplier coder and the multiplier ?? obtained by decoding the multiplier ? is outputted. The process is performed for each frame.
    Type: Application
    Filed: March 16, 2011
    Publication date: July 7, 2011
    Applicants: NIPPON TELEGRAPH AND TELEPHONE CORPORATION, The University of Tokyo
    Inventors: Takehiro Moriya, Noboru Harada, Yutaka Kamamoto, Takuya Nishimoto, Shigeki Sagayama
  • Publication number: 20110071822
    Abstract: Certain aspects relate to providing an at least one audio source to at least one user. Certain aspects relate to selectively modifying an at least one first sound source to be provided to the at least one user, wherein the at least one first sound source is combined with an at least one second sound source, and wherein the selectively modifying is performed relative to the at least one audio source based at least in part on at least some specific information of the at least one first sound source. Other aspects relate to selectively modifying the at least one first sound source to be provided to the at least one user relative to the at least one second sound source based at least in part on at least some specific information of the at least one first sound source.
    Type: Application
    Filed: September 22, 2010
    Publication date: March 24, 2011
    Inventors: Alexander J. Cohen, Edward K.Y. Jung, Royce A. Levien, Robert W. Lord, Mark A. Malamud, William Henry Mangione-Smith, John D. Rinaldo, JR., Clarence T. Tegreene, Lowell L. Wood, JR.
  • Publication number: 20110060593
    Abstract: An output circuit for audio codec chip includes a noise eliminating circuit electrically coupled to the audio codec chip for eliminating noise signals. The noise eliminating circuit includes a first switch and a second switch. When the audio codec chip output signals jump from low voltage level to high voltage level, the noise signals are grounded via the first and second switches respectively.
    Type: Application
    Filed: November 13, 2009
    Publication date: March 10, 2011
    Applicants: HONG FU JIN PRECISION INDUSTRY (ShenZhen) CO., LTD ., HON HAI PRECISION INDUSTRY CO., LTD.
    Inventor: KE-YOU HU
  • Publication number: 20100324892
    Abstract: A code excited linear prediction type speech coder, which includes a seed storage that stores seeds used as an initial state of oscillation, and an oscillator that generates different vector sequences in accordance with values of the seeds stored in the seed storage and outputs the vector sequences as excitation vectors. The speech coder also includes a linear predictive coding synthesis filter that receives, as input, the excitation vectors, which are the vector sequences generated in accordance with the values of the seeds, that synthesizes the excitation vectors, and that outputs a synthesized speech.
    Type: Application
    Filed: August 27, 2010
    Publication date: December 23, 2010
    Applicant: Panasonic Corporation
    Inventors: Kazutoshi YASUNAGA, Toshiyuki MORII, Taisuke WATANABE
  • Publication number: 20100305953
    Abstract: A method of generating a frame of audio data for an audio signal from preceding audio data for the audio signal that precede the frame of audio data, the method comprising the steps of: predicting a predetermined number of data samples for the frame of audio data based on the preceding audio data, to form predicted data samples; identifying a section of the preceding audio data for use in generating the frame of audio data; and forming the audio data of the frame of audio data as a repetition (602) of at least part of the identified section to span the frame of audio data, wherein the beginning of the frame of audio data comprises a combination of a subset of the repetition (602) of the at least part of the identified section and the predicted data samples.
    Type: Application
    Filed: May 14, 2007
    Publication date: December 2, 2010
    Applicant: Freescale Semiconductor, Inc.
    Inventors: Adrian Susan, Mihai Neghina
  • Publication number: 20100305955
    Abstract: The present invention relates to encoding technology. The encoding method includes selecting a second encoding mode for encoding an input frame signal according to an analysis on signal characteristic of the input frame signal; obtaining coding demand values for a preset first encoding mode and the second encoding mode which are used to encode the input frame signal; determining, from the above encoding modes based on the coding demand values, an encoding mode for encoding the input frame signal; and multiplexing information of the determined encoding mode and encoded data which are encoded according to the determined encoding mode. Hence, the compatibility and the prioritization in terms of the encoding modes can be achieved.
    Type: Application
    Filed: May 28, 2010
    Publication date: December 2, 2010
    Applicant: HUAWEI TECHNOLOGIES CO., LTD.
    Inventors: Lei MIAO, Fengyan QI, Qing ZHANG
  • Publication number: 20100262421
    Abstract: Provided is an encoding device which improves the sound quality of a stereo signal while maintaining a low bit rate. The encoding device includes: an LP inverse filter (121) which LP-inverse-filterS a left signal L(n) by using an inverse quantization linear prediction coefficient AdM(z) of a monaural signal; a T/F conversion unit (122) which converts the left sound source signal Le(n) from a temporal region to a frequency region; an inverse quantizer (123) which inverse-quantizes encoded information Mqe; spectrum division units (124, 125) which divide a high-frequency component of the sound source signal Mde(f) and the left signal Le(f) into a plurality of bands; and scale factor calculation units (126, 127) which calculate scale factors ai and ssi by using a monaural sound source signal Mdeh,i(f), a left sound source signal Leh,i(f), Mdeh,i(f), and right sound source signal Reh,i(f) of each divided band.
    Type: Application
    Filed: November 4, 2008
    Publication date: October 14, 2010
    Applicant: PANASONIC CORPORATION
    Inventors: Kok Seng Chong, Koji Yoshida, Masahiro Oshikiri
  • Publication number: 20100235172
    Abstract: A method for processing an audio signal, comprising: receiving the audio signal; and processing the received audio signal, wherein the audio signal is processed according to a scheme comprising: comparing a size information of at least two blocks of A+1 level with a size information of a block of A level corresponding to the at least two of A+1 level; and, determining the at least two blocks of A+1 level as an optimum block if the size information of the at least two blocks of A+1 level is less than the size information of the block of A level is disclosed.
    Type: Application
    Filed: December 6, 2007
    Publication date: September 16, 2010
    Inventor: Tilman Liebchen
  • Publication number: 20100223054
    Abstract: A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame.
    Type: Application
    Filed: May 14, 2010
    Publication date: September 2, 2010
    Applicant: BROADCOM CORPORATION
    Inventors: Elias Nemer, Wilfrid LeBlanc, Syavosh Zad-Issa, Jes Thyssen
  • Publication number: 20100198586
    Abstract: A processed representation of an audio signal having a sequence of frames is generated by sampling the audio signal within first and second frames of the sequence of frames, the second frame following the first frame, the sampling using information on a pitch contour of the first and second frames to derive a first sampled representation. The audio signal is sampled within the second and third frames, the third frame following the second frame in the sequence of frames. The sampling uses the information on the pitch contour of the second frame and information on a pitch contour of the third frame to derive a second sampled representation. A first scaling window is derived for the first sampled representation, and a second scaling window is derived for the second sampled representation, the scaling windows depending on the samplings applied to derive the first sampled representations or the second sampled representation.
    Type: Application
    Filed: March 23, 2009
    Publication date: August 5, 2010
    Applicant: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e. V.
    Inventors: Bernd Edler, Sascha Disch, Ralf Geiger, Stefan Bayer, Ulrich Kraemer, Guillaume Fuchs, Max Neuendorf, Markus Multrus, Gerald Schuller, Harald Popp
  • Publication number: 20100174534
    Abstract: A method of encoding speech, the method comprising: receiving a signal representative of speech to be encoded; at each of a plurality of intervals during the encoding, determining a pitch lag between portions of the signal having a degree of repetition; selecting for a set of said intervals a pitch lag vector from a pitch lag codebook of such vectors, each pitch lag vector comprising a set of offsets corresponding to the offset between the pitch lag determined for each said interval and an average pitch lag for said set of intervals, and transmitting an indication of the selected vector and said average over a transmission medium as part of the encoded signal representative of said speech.
    Type: Application
    Filed: June 5, 2009
    Publication date: July 8, 2010
    Inventor: Koen Bernard Vos
  • Publication number: 20100174532
    Abstract: A method, system and program for encoding and decoding speech according to a source-filter model whereby speech is modelled to comprise a source signal filtered by a time-varying filter. The method comprises: receiving a speech signal comprising successive frames, for each of a plurality of frames of the speech signal, deriving a first line spectral frequency vector for a first portion of the frame, and a second line spectral frequency vector for a second portion of the frame, and determining a transmit line spectral frequency vector and an interpolation factor based on the first and second line spectral frequency vectors, and on the transmit line spectral frequency vector for a preceding one of the frames.
    Type: Application
    Filed: June 5, 2009
    Publication date: July 8, 2010
    Inventors: Koen Bernard Vos, Karsten Vandborg Sorensen, Soren Skak Jensen
  • Publication number: 20100153121
    Abstract: An information coding apparatus includes a predictive signal generator that generates a predictive signal; a predictive residual signal generator that generates a predictive residual signal; a quantizer that quantizes a quantization input signal generated based on the predictive residual signal; a quantization error signal generator that generates a quantization error signal; a feedback signal generator that generates a feedback signal for controlling the frequency characteristic of the quantization noise after decoding based on the quantization error signal; and a quantization input signal generator that generates the quantization input signal.
    Type: Application
    Filed: December 16, 2009
    Publication date: June 17, 2010
    Inventors: Yasuhiro TOGURI, Jun Matsumoto
  • Publication number: 20100153100
    Abstract: An address generator for searching an algebraic codebook is disclosed. The address generator includes: a multiplier multiplying the dimension and a width value of a correlation matrix; a first adder adding a length value and an offset address of the correlation matrix; and a second adder adding the results of the multiplier and the first adder to generate an address for algebraic codebook searching. The amount of calculation required for an address calculation to search an algebraic codebook can be reduced.
    Type: Application
    Filed: September 3, 2009
    Publication date: June 17, 2010
    Applicant: Electronics and Telecommunications Research Institute
    Inventors: Byun Kyung Jin, Koo Bon Tae, Eum Nak Woong
  • Publication number: 20100125454
    Abstract: Packet loss concealment systems and methods are described that may be used in conjunction with a Bluetooth® Low-Complexity Sub-band Coding (LC-SBC) codec or other sub-band codecs, including but not limited to an MPEG-1 Audio Layer 3 (MP3) codec, an Advanced Audio Coding (AAC) codec, and a Dolby AC-3 codec.
    Type: Application
    Filed: November 6, 2009
    Publication date: May 20, 2010
    Applicant: BROADCOM CORPORATION
    Inventors: Robert W. Zopf, Laurent Pilati
  • Publication number: 20100121632
    Abstract: Provided is a stereo audio encoding device which can improve the ICP (Inter-channel Prediction) performance of a stereo audio signal while suppressing the bit rate. The device (100) includes: a QMF analysis unit (101) which divides two channel signals constituting a stereo audio signal into a plurality of frequency band signals; a monaural signal generation unit (104) which generates a monaural signal by averaging the two channel signals of the divided frequency bands; parameter band constituting units (102, 105) each of which collects one or more of the continuous frequency bands to constitute a parameter band in such a manner that less bands are contained in a lower frequency for the two channel signals and monaural signals of the divided frequency bands; and an ICP analysis unit (106) which performs inter-channel prediction by using the channel signal and the monaural signal of the divided frequency bands.
    Type: Application
    Filed: April 24, 2008
    Publication date: May 13, 2010
    Applicant: PANASONIC CORPORATION
    Inventor: Kok Seng Chong
  • Publication number: 20100114566
    Abstract: An apparatus and method for encoding/decoding a speech signal which determines a variable bit rate based on reserved bits obtained from a target bit rate, is provided. The variable bit rate is determined based on a source feature of the speech signal and the reserved bits is obtained based on the target bit rate. The apparatus for encoding the speech signal may include a linear predictive (LP) analysis unit/quantization unit to determine an immittance spectral frequencies (ISF) index, a closed loop pitch search unit, a fixed codebook search unit, a gain vector quantization (VQ) unit to determine a gain vector quantization (VQ) index, and a bit rate control unit to control at least two indexes of the ISF index, the pitch index, the code index, and the gain VQ index to be encoded to be variable bit rates based on a source feature of a speech signal and the reserved bits.
    Type: Application
    Filed: July 28, 2009
    Publication date: May 6, 2010
    Applicant: SAMSUNG ELECTRONICS CO., LTD.
    Inventors: Ho Sang Sung, Eun Mi Oh
  • Publication number: 20100082337
    Abstract: Disclosed is an adaptive sound source vector quantization device capable of improving quantization accuracy of adaptive sound source vector quantization while suppressing increase of the calculation amount in CELP sound encoding which performs encoding in sub-frame unit.
    Type: Application
    Filed: December 14, 2007
    Publication date: April 1, 2010
    Applicant: PANASONIC CORPORATION
    Inventors: Kaoru Sato, Toshiyuki Morii
  • Publication number: 20100082335
    Abstract: The system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, a transmitter analysis filter for receiving the digital speech signal and dividing it into a baseband signal and an enhancement residual band signal, a standard baseband encoder for accepting the baseband signal and coding it using an ITU-T encoder, an additional baseband encoder for reducing standard coding distortion in the baseband signal, an enhancement residual band encoder for coding a signal obtained by removing the coded baseband signal from the original digital speech signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.
    Type: Application
    Filed: December 4, 2009
    Publication date: April 1, 2010
    Applicant: Electronics and Telecommunications Research Institute
    Inventors: Ho-Sang SUNG, Dae-Hwan HWANG, Dae-Hee YOUN, Hong-Goo KANG, Young-Cheol PARK, Ki-Seung LEE, Sung-Kyo JUNG, Kyung-Tae KIM
  • Publication number: 20100057473
    Abstract: Aspects of a method and system for dual voice path processing in an audio CODEC may enable selecting two or more signals received via one or more audio input devices, and filtering and down-sampling each of the selected signals via two or more signal processing branches. Furthermore, an output sample rate of each of the signal processing branches may be configured independently. The signal processing branches may each comprise one or more IIR filters with configurable coefficients and one or more cascaded-integrate-comb (CIC) decimation filters having a configurable decimation ratio. A first of the selected signals, may be filtered and/or down-sampled to generate a signal having a first, lower, sample rate, and a second of the signals may be filtered and/or down-sampled to generate a signal having a second, higher, sample rate. One or more post-processing algorithms such as audio beamforming may also be performed on the selected signals.
    Type: Application
    Filed: October 9, 2008
    Publication date: March 4, 2010
    Inventors: Hongwei Kong, Thirunathan Sutharsan
  • Publication number: 20100049506
    Abstract: A method, device and system to implement hiding the loss packet are provided. The provided method, device and system recover the lost frame according to the data before and after the lost frame and enhances the correlation of the recovered lost frame data and the data after the lost frame. A method and device for estimating pitch period are also provided which select a pitch period from the initial pitch period and the pitch periods corresponding to the frequencies which are one or more times higher than the frequencies corresponding to the initial pitch period as the final estimated pitch period, may improve frequency multiplication when estimating the pitch period; in addition, by tuning of the pitch period by matching the waves, the error of estimating pitch period may be reduced and the quality of the audio data may be improved.
    Type: Application
    Filed: November 2, 2009
    Publication date: February 25, 2010
    Inventors: Wuzhou Zhan, Dongqi Wang
  • Publication number: 20100049510
    Abstract: A method, device and system to implement hiding the loss packet are provided. The provided method, device and system recover the lost frame according to the data before and after the lost frame and enhances the correlation of the recovered lost frame data and the data after the lost frame. A method and device for estimating pitch period are also provided which select a pitch period from the initial pitch period and the pitch periods corresponding to the frequencies which are one or more times higher than the frequencies corresponding to the initial pitch period as the final estimated pitch period, may improve frequency multiplication when estimating the pitch period; in addition, by tuning of the pitch period by matching the waves, the error of estimating pitch period may be reduced and the quality of the audio data may be improved.
    Type: Application
    Filed: November 2, 2009
    Publication date: February 25, 2010
    Inventors: Wuzhou Zhan, Dongqi Wang
  • Publication number: 20100049508
    Abstract: Provided is an audio encoding device which performs a closed loop search of a gain and a sound source vector without significantly increasing the calculation amount as compared to an open loop search. In the audio encoding device, firstly, a first parameter decision unit (121) performs a sound source search by an adaptive sound source codebook and then a second parameter decision unit (122) simultaneously performs by a closed loop, the sound source and the gain search by using a fixed sound source codebook. More specifically, for a combination of a fixed sound source vector and gain, the sum of a value obtained by multiplying a candidate fixed sound source vector by a candidate gain and a value obtained by multiplying an adaptive sound source vector by a candidate gain is subjected to a combination filter formed by a filter coefficient based on a quantization linear prediction coefficient so as to generate a combined signal.
    Type: Application
    Filed: December 14, 2007
    Publication date: February 25, 2010
    Applicant: PANASONIC CORPORATION
    Inventor: Toshiyuki Morii
  • Publication number: 20100049509
    Abstract: Disclosed are an audio encoding device and an audio decoding device which reduce degradation of subjective quality of a decoding signal caused by power mismatch of a decoding signal which is generated by a concealing process upon disappearance of a frame. When a frame is lost, a past encoding parameter is used to obtain a concealed LPC of the current frame and a concealed sound source parameter. A normal CELP decoding is performed from the obtained concealed sound source parameter. Correction is performed by using a conceal parameter on the obtained concealed LPC and the concealed sound source signal. The power of the corrected concealed sound source signal is adjusted to match a reference sound source power. A filter gain of the synthesis filter is adjusted so as to adjust the power of a decoded sound signal to the power of a decoded sound signal during an error-free state.
    Type: Application
    Filed: February 29, 2008
    Publication date: February 25, 2010
    Applicant: PANASONIC CORPORATION
    Inventors: Takuya Kawashima, Hiroyuki Ehara, Koji Yoshida
  • Publication number: 20100049507
    Abstract: An apparatus for noise suppression having a linear prediction analysis circuit having an LP error filter (LFF), which takes a first, noisy voice signal y(n)=x(n)+?(n) as a basis for producing an LP-error-filter output signal e(n), having a coefficient calculation unit (KBE), which updates the coefficients of the LP error filter on the basis of the internal signals (including the input and out signals y(n) and e(n)) in the LP error filter, and having a subtraction unit, which subtracts the LP error filter output signal e(n) from the first voice signal y(n) in a subtractor and, following the subtraction, outputs the remainder as a second voice signal x(n)=y(n)?e(n) in which the noise is suppressed.
    Type: Application
    Filed: September 6, 2007
    Publication date: February 25, 2010
    Applicants: Technische Universitat Graz, Forschungsholding TU Graz GmbH
    Inventors: Erhard Rank, Gernot Kubin
  • Publication number: 20100023325
    Abstract: A device and a method for quantizing a LPC filter in the form of an input vector in a quantization domain, comprises a calculator of a first-stage approximation of the input vector, a subtractor of the first-stage approximation from the input vector to produce a residual vector, a calculator of a weighting function from the first-stage approximation, a warper of the residual vector with the weighting function, and a quantizer of the weighted residual vector to supply a quantized weighted residual vector.
    Type: Application
    Filed: July 10, 2009
    Publication date: January 28, 2010
    Applicant: VOICEAGE CORPORATION
    Inventors: Bruno BESSETTE, Philippe Gournay, Redwan Salami
  • Publication number: 20100017197
    Abstract: It is an object to disclose a voice coding device, etc. in which the deterioration of a voice quality of a decoded signal can be reduced in the case that low frequency domain components of a spectrum are used for coding high frequency domain components and that no low frequency domain components exist.
    Type: Application
    Filed: November 1, 2007
    Publication date: January 21, 2010
    Applicant: PANASONIC CORPORATION
    Inventor: Masahiro Oshikiri
  • Publication number: 20100010811
    Abstract: Disclosed is a stereo audio encoding device capable of reducing a bit rate. In this device, a stereo audio encoding unit (103) performs LPC analysis on an L channel signal and an R channel signal so as to obtain an L channel LPC coefficient and an R channel LPC coefficient. An LPC coefficient adaptive filter (105) obtains an LPC coefficient adaptive filter parameter to minimize the mean square error between the L channel LPC coefficient and the R channel LPC coefficient. An LPC coefficient reconfiguration unit (106) reconfigures the R channel LPC coefficient by using the L channel LPC coefficient and the LPC coefficient adaptive filter parameter. A route calculation unit (107) calculates a polynomial route indicating the safety of the R channel reconfigured LPC coefficient. A selection unit (108) selects and outputs the LPC coefficient adaptive filter parameter or the R channel LPC coefficient according to the safety of the R channel reconfigured LPC coefficient.
    Type: Application
    Filed: August 2, 2007
    Publication date: January 14, 2010
    Applicant: PANASONIC CORPORATION
    Inventors: Jiong Zhou, Sua Hong Neo, Koji Yoshida, Michiyo Goto