Interpolation Patents (Class 704/265)
  • Patent number: 8165882
    Abstract: Apparatus and method for generating high quality synthesized speech having smooth waveform concatenation. The apparatus includes a pitch frequency calculation section, a pitch synchronization position calculation section, a unit waveform storage, a unit waveform selection section, a unit waveform generation section, and a waveform synthesis section. The unit waveform generation section includes a conversion ratio calculation section, a sampling rate conversion section, and a unit waveform re-selection section. The conversion ratio calculation section calculates a sampling rate conversion ratio from the pitch information and the position of pitch synchronization, and the sampling rate conversion section converts the sampling rate of the unit waveform, delivered as input, based on the sampling rate conversion ratio.
    Type: Grant
    Filed: September 4, 2006
    Date of Patent: April 24, 2012
    Assignee: NEC Corporation
    Inventors: Masanori Kato, Satoshi Tsukada
  • Patent number: 8145477
    Abstract: Systems, methods, and apparatus described include waveform alignment operations in which a single set of evaluated cosines and sines is used to calculate cross-correlations of two periodic waveforms at two different phase shifts.
    Type: Grant
    Filed: December 1, 2006
    Date of Patent: March 27, 2012
    Inventors: Sharath Manjunath, Ananthapadmanabhan A. Kandhadai
  • Publication number: 20120065980
    Abstract: An electronic device for coding a transient frame is described. The electronic device includes a processor and executable instructions stored in memory that is in electronic communication with the processor. The electronic device obtains a current transient frame. The electronic device also obtains a residual signal based on the current transient frame. Additionally, the electronic device determines a set of peak locations based on the residual signal. The electronic device further determines whether to use a first coding mode or a second coding mode for coding the current transient frame based on at least the set of peak locations. The electronic device also synthesizes an excitation based on the first coding mode if the first coding mode is determined. The electronic device also synthesizes an excitation based on the second coding mode if the second coding mode is determined.
    Type: Application
    Filed: September 8, 2011
    Publication date: March 15, 2012
    Applicant: QUALCOMM Incorporated
    Inventors: Venkatesh Krishnan, Ananthapadmanabhan Arasanipalai Kandhadai
  • Patent number: 8126578
    Abstract: A method and system for optimally repairing a clipped audio signal. Clipping occurs when a waveform exceeds a dynamic range of a recording device. Portions of an audio signal exceeding the dynamic range or saturation level of the recording device are clipped, causing distortion when the clipped recorded signal is played. To address this problem, successive frames of the clipped audio data are repaired to fill in gaps where the data were clipped. For each frame, an iterative process repetitively estimates an auto-covariance and detects clipped samples in the frame or a sub-frame in order to compute a least-squares solution for the frame that interpolates the clipped data. The process can cause inverted peaks in the repaired data, which must then be rectified to produced corrected repaired data. The corrected repaired data for the successive frames are recombined using interpolation, to produce a complete repaired audio data set.
    Type: Grant
    Filed: September 26, 2007
    Date of Patent: February 28, 2012
    Assignee: University of Washington
    Inventors: Les Atlas, Charles Pascal Clark
  • Patent number: 8126162
    Abstract: An audio signal interpolation apparatus is configured to perform interpolation processing on the basis of audio signals preceding and/or following a predetermined segment on a time axis so as to obtain an audio signal corresponding to the predetermined segment. The audio signal interpolation apparatus includes a waveform formation unit configured to form a waveform for the predetermined segment on the basis of time-domain samples of the preceding and/or the following audio signals and a power control unit configured to control power of the waveform for the predetermined segment formed by the waveform formation unit using a non-linear model selected on the basis of the preceding audio signal when the power of the preceding audio signal is larger than that of the following audio signal, or the following audio signal when the power of the preceding audio signal is smaller than that of the following audio signal.
    Type: Grant
    Filed: May 23, 2007
    Date of Patent: February 28, 2012
    Assignee: Sony Corporation
    Inventors: Chunmao Zhang, Toru Chinen
  • Patent number: 8121832
    Abstract: Provided are a method and apparatus for encoding and decoding a high frequency signal by using a low frequency signal. The high frequency signal can be encoded by extracting a coefficient by linear predicting a high frequency signal, and encoding the coefficient, generating a signal by using the extracted coefficient and a low frequency signal, and encoding the high frequency signal by calculating a ratio between the high frequency signal and an energy value of the generated signal. Also, the high frequency signal can be decoded by decoding a coefficient, which is extracted by linear predicting a high frequency signal, and a low frequency signal, and generating a signal by using the decoded coefficient and the decoded low frequency signal, and adjusting the generated signal by decoding a ratio between the generated signal and an energy value of the high frequency signal.
    Type: Grant
    Filed: November 15, 2007
    Date of Patent: February 21, 2012
    Assignee: Samsung Electronics Co., Ltd.
    Inventors: Ki-hyun Choo, Lei Miao, Eun-mi Oh
  • Patent number: 8065141
    Abstract: A signal processing apparatus includes a decoding unit, an analyzing unit, a synthesizing unit, and a selecting unit. The decoding unit decodes an input encoded audio signal and outputs a playback audio signal. When loss of the encoded audio signal occurs, the analyzing unit analyzes the playback audio signal output before the loss occurs and generates a linear predictive residual signal. The synthesizing unit synthesizes a synthesized audio signal on the basis of the linear predictive residual signal. The selecting unit selects one of the synthesized audio signal and the playback audio signal and outputs the selected audio signal as a continuous output audio signal.
    Type: Grant
    Filed: August 24, 2007
    Date of Patent: November 22, 2011
    Assignee: Sony Corporation
    Inventor: Yuuji Maeda
  • Patent number: 8015012
    Abstract: Portions from segment boundary regions of a plurality of speech segments are extracted. Each segment boundary region is based on a corresponding initial unit boundary. Feature vectors that represent the portions in a vector space are created. For each of a plurality of potential unit boundaries within each segment boundary region, an average discontinuity based on distances between the feature vectors is determined. For each segment, the potential unit boundary associated with a minimum average discontinuity is selected as a new unit boundary.
    Type: Grant
    Filed: July 28, 2008
    Date of Patent: September 6, 2011
    Assignee: Apple Inc.
    Inventor: Jerome R. Bellegarda
  • Patent number: 8010355
    Abstract: A method of reducing noise in a speech signal involves converting the speech signal to the frequency domain using a fast fourier transform (FFT), creating a subset of selected spectral subbands, determining the appropriate gain for each subband, and interpolating the gains to match the number of FFT points. The converted speech signal is then filtered using the interpolated gains as filter coefficients, and an inverse FFT performed on the processed signal to recover the time domain output signal.
    Type: Grant
    Filed: April 25, 2007
    Date of Patent: August 30, 2011
    Assignee: Zarlink Semiconductor Inc.
    Inventor: Kamran Rahbar
  • Patent number: 8010362
    Abstract: A voice conversion rule and a rule selection parameter are stored. The voice conversion rule converts a spectral parameter vector of a source speaker to a spectral parameter vector of a target speaker. The rule selection parameter represents the spectral parameter vector of the source speaker. A first voice conversion rule of start time and a second voice conversion rule of end time in a speech unit of the source speaker are selected by the spectral parameter vector of the start time and the end time. An interpolation coefficient corresponding to the spectral parameter vector of each time in the speech unit is calculated by the first voice conversion rule and the second voice conversion rule. A third voice conversion rule corresponding to the spectral parameter vector of each time in the speech unit is calculated by interpolating the first voice conversion rule and the second voice conversion rule with the interpolation coefficient.
    Type: Grant
    Filed: January 22, 2008
    Date of Patent: August 30, 2011
    Assignee: Kabushiki Kaisha Toshiba
    Inventors: Masatsune Tamura, Takehiro Kagoshima
  • Patent number: 7957973
    Abstract: An audio signal interpolation device comprises a spectral movement calculation unit which determines a spectral movement which is indicative of a difference in each of spectral components between a frequency spectrum of a current frame of an input audio signal and a frequency spectrum of a previous frame of the input audio signal stored in a spectrum storing unit. An interpolation band determination unit determines a frequency band to be interpolated by using the frequency spectrum of the current frame and the spectral movement. A spectrum interpolation unit performs interpolation of spectral components in the frequency band for the current frame by using either the frequency spectrum of the current frame or the frequency spectrum of the previous frame.
    Type: Grant
    Filed: July 25, 2007
    Date of Patent: June 7, 2011
    Assignee: Fujitsu Limited
    Inventors: Masakiyo Tanaka, Masanao Suzuki, Miyuki Shirakawa, Takashi Makiuchi
  • Patent number: 7945446
    Abstract: Spectrum envelope of an input sound is detected. In the meantime, a converting spectrum is acquired which is a frequency spectrum of a converting sound comprising a plurality of sounds, such as unison sounds. Output spectrum is generated by imparting the detected spectrum envelope of the input sound to the acquired converting spectrum. Sound signal is synthesized on the basis of the generated output spectrum. Further, a pitch of the input sound may be detected, and frequencies of peaks in the acquired converting spectrum may be varied in accordance with the detected pitch of the input sound. In this manner, the output spectrum can have the pitch and spectrum envelope of the input sound and spectrum frequency components of the converting sound comprising a plurality of sounds, and thus, unison sounds can be readily generated with simple arrangements.
    Type: Grant
    Filed: March 9, 2006
    Date of Patent: May 17, 2011
    Assignee: Yamaha Corporation
    Inventors: Hideki Kemmochi, Yasuo Yoshioka, Jordi Bonada
  • Publication number: 20110077945
    Abstract: This invention relates to a method, a computer program product, apparatuses and a system for extracting coded parameter set from an encoded audio/speech stream, said audio/speech stream being distributed to a sequence of packets, and generating a time scaled encoded audio/speech stream in the parameter coded domain using said extracted coded parameter set.
    Type: Application
    Filed: June 6, 2007
    Publication date: March 31, 2011
    Applicant: NOKIA CORPORATION
    Inventors: Pasi Sakari Ojala, Ari Kalevi Lakaniemi
  • Patent number: 7899667
    Abstract: A waveform interpolation speech coding apparatus and method for reducing complexity thereof are disclosed. The waveform interpolation speech coding apparatus includes: a waveform interpolation encoding unit for receiving a speech signal, calculating parameters for a waveform interpolation from the received speech signal, and quantizing the calculating parameters; and a realignment parameter calculating unit for restoring a characteristic waveform (CW) using the quantized parameter, calculating a realignment parameter that maximizes a cross-correlation among consecutive CWs for the restored CW.
    Type: Grant
    Filed: December 19, 2006
    Date of Patent: March 1, 2011
    Assignee: Electronics and Telecommunications Research Institute
    Inventors: Kyung-Jin Byun, Ik-Soo Eo, Hee-Bum Jung, Nak-Woong Eum
  • Publication number: 20100228550
    Abstract: An audio signal interpolation method includes: a step of inputting an audio signal in which a higher range component has been cut off; a step of dividing the inputted audio signal into an in-phase component signal and a differential phase component signal; a step of combining the in-phase component signal and a differential phase component signal having a high range component interpolated; a step of performing a high-pass filtering on the combined audio signal and outputting the audio signal formed by a high range component; a step of delaying the inputted audio signal by a time corresponding to a phase delay caused by the interpolation; and a step of adding the delayed audio signal to the audio signal subjected to the high-pass filtering.
    Type: Application
    Filed: September 29, 2008
    Publication date: September 9, 2010
    Applicant: D&M HOLDINGS INC.
    Inventors: Masaki Matsuoka, Shigeki Namiki
  • Patent number: 7765100
    Abstract: A method and an apparatus for recovering a line spectrum pair (LSP) parameter of a spectrum region when frame loss occurs during speech decoding and a speech decoding apparatus adopting the same are provided. The method of recovering an LSP parameter in speech decoding includes: if it is determined that a received speech packet has an erased frame, converting an LSP parameter of a previous good frame (PGF) of the erased frame or LSP parameters of the PGF and a next good frame (NGF) of the erased frame into a spectrum region and obtaining a spectrum envelope of the PGF or spectrum envelopes of the PGF and NGF; recovering a spectrum envelope of the erased frame using the spectrum envelope of the PGF or the spectrum envelopes of the PGF and NGF; and converting the recovered spectrum envelope of the erased frame into an LSP parameter of the erased frame.
    Type: Grant
    Filed: February 6, 2006
    Date of Patent: July 27, 2010
    Assignee: Samsung Electronics Co., Ltd.
    Inventors: Hosang Sung, Seungho Choi, Kihyun Choo
  • Patent number: 7684978
    Abstract: The present invention overcomes problems of tandem coding method such as degradation of speech quality, increased system latency and computations. An apparatus for trans-coding between code excited linear prediction (CELP) type codecs with different bandwidths, includes: a format parameter translating unit for generating output formant parameters by translating formant parameters from input CELP format to output CELP format; a formant parameter quantizing unit for receiving the output format formant parameters and quantizing the output format formant filter coefficients; an excited parameter translating unit for generating output excitation parameters by translating excitation parameters from input CELP format to output CELP format; and an excitation quantizing unit for receiving the output format excitation parameters and quantizing the output format excitation parameters.
    Type: Grant
    Filed: October 30, 2003
    Date of Patent: March 23, 2010
    Assignee: Electronics and Telecommunications Research Institute
    Inventors: Jongmo Sung, Sang Taick Park, Do Young Kim, Bong Tae Kim
  • Patent number: 7680665
    Abstract: A frequency interpolating device for restoring a signal similar to the original signal by creating a suppressed frequency component of a specific frequency band of the original signal, approximately from the input signal having the suppressed frequency component. In the frequency interpolating device, when the suppressed frequency component is artificially created from the input signal and added to the input signal, the additional level is set dynamically and adaptively on the basis of the spectrum pattern of the remaining frequency component of the input signal. This setting of the addition level is done by searching a look-up table which stores data that causes a plurality of reference frequency spectrum patterns to be associated with predetermined addition levels. Moreover, the data stored in the table is created on the basis of the results of either an aural test on a plurality of signal sample sounds or a physical frequency analysis on the massive signal data.
    Type: Grant
    Filed: August 24, 2001
    Date of Patent: March 16, 2010
    Assignee: Kabushiki Kaisha Kenwood
    Inventors: Norihisa Shigyo, Norikazu Tanaka
  • Patent number: 7676361
    Abstract: A voice signal interpolation apparatus is provided which can restore original human voices from human voices in a compressed state while maintaining a high sound quality. When a voice signal representative of a voice to be interpolated is acquired by a voice data input unit 1, a pitch deriving unit 2 filters this voice signal to identify a pitch length from the filtering result. A pitch length fixing unit 3 makes the voice signal have a constant time length of a section corresponding to a unit pitch, and generates pitch waveform data. A sub-band dividing unit 4 converts the pitch waveform data into sub-band data representative of a spectrum. A plurality of sub-band data pieces are averaged by an averaging unit 5 and thereafter a sub-band synthesizing unit 6 converts the sub-band data pieces into a signal representative of a waveform of the voice by a sub-band synthesizing unit 6.
    Type: Grant
    Filed: May 7, 2007
    Date of Patent: March 9, 2010
    Assignee: Kabushiki Kaisha Kenwood
    Inventor: Yasushi Sato
  • Patent number: 7643996
    Abstract: An Enhanced analysis-by-synthesis Waveform Interpolative speech coder able to operate at 4 kbps. Novel features include analysis-by-synthesis quantization of the slowly evolving waveform, analysis-by-synthesis vector quantization of the dispersion phase, a special pitch search for transitions, and switched-predictive analysis-by-synthesis gain vector quantization. Subjective quality tests indicate that it exceeds MPEG-4 at 4 kbps and of G.723.1 at 6.3 kbps.
    Type: Grant
    Filed: December 1, 1999
    Date of Patent: January 5, 2010
    Assignee: The Regents of the University of California
    Inventor: Oded Gottesman
  • Publication number: 20090326950
    Abstract: A voice waveform interpolating apparatus for interpolating part of stored voice data by another part of the voice data so as to generate voice data. To achieve this, it comprises a voice storage unit, an interpolated waveform generation unit generating interpolated voice data, and a waveform combining unit outputting voice data, a part of the voice data is replaced with another part of the voice data, and further comprises an interpolated waveform setting function unit judging if the other part of the voice data is appropriate as interpolated voice data to be generated by the interpolated waveform generation unit.
    Type: Application
    Filed: August 31, 2009
    Publication date: December 31, 2009
    Applicant: FUJITSU LIMITED
    Inventor: Chikako Matsumoto
  • Patent number: 7636662
    Abstract: A system and method is provided for synthesizing audio-visual content in a video image processor. A content synthesis application processor extracts audio features and video features from audio-visual input signals that represent a speaker who is speaking. The processor uses the extracted visual features to create a computer generated animated version of the face of the speaker. The processor synchronizes facial movements of the animated version of the face of the speaker with a plurality of audio logical units such as phonemes that represent the speaker's speech. In this manner the processor synthesizes an audio-visual representation of the speaker's face that is properly synchronized with the speaker's speech.
    Type: Grant
    Filed: September 28, 2004
    Date of Patent: December 22, 2009
    Assignee: Koninklijke Philips Electronics N.V.
    Inventors: Nevenka Dimtrova, Andrew Miller, Dongge Li
  • Patent number: 7627198
    Abstract: Standard Definition signals are divided into blocks at a tap extracting unit, and pixel data contained in each block is extracted as a class tap. A class classification unit obtains class code based on the pixel data contained in the class tap. An auxiliary data generating unit generates auxiliary data regarding conversion into High Definition signals, based on the class tap extracted by the tap extracting unit. A data generation processing unit performs processing based on the class code and the auxiliary data, thereby yielding excellent High Definition signals.
    Type: Grant
    Filed: May 25, 2004
    Date of Patent: December 1, 2009
    Assignee: Sony Corporation
    Inventors: Tetsujiro Kondo, Takashi Nakanishi, Daisuke Kikuchi, Shizuo Chikaoka, Takeshi Miyai, Yoshiaki Nakamura, Tsugihiko Haga
  • Patent number: 7606711
    Abstract: The pitch extracting part generates a pitch waveform signal in a manner making the time interval of the pitch of the input audio sound data to be the same. After the number of samples in each region is made to be the same by the re-sampling part, the pitch waveform signal is changed into a subband data that express a time-varying-strength of a basic frequency composition and a higher harmonic composition by the subband analyzing part. The subband data are superimposed by a modulation wave composition that expresses attaching data of an attaching object by the data attaching part and is regarded as a bit stream to output through a nonlinear quantizing. A portion expressing the higher harmonic composition that is made corresponding to the audio sound expressed by this audio sound data in the subband data are deleted by the encoding part.
    Type: Grant
    Filed: September 22, 2006
    Date of Patent: October 20, 2009
    Assignee: Kenwood Corporation
    Inventor: Yasushi Sato
  • Patent number: 7596497
    Abstract: A speech synthesis apparatus and a speech synthesis method, in which a waveform of a desired formant shape may be generated with a small volume of computing operations. A voiced sound generating unit of the speech synthesis apparatus includes n single formant generating units, an adder for summing these outputs to generate a one-pitch waveform, a one-pitch buffer unit, and a waveform overlapping unit for overlapping a number of the one-pitch waveforms as the one-pitch waveform is shifted by one pitch period each time. Each single formant generating unit is supplied with three parameters, namely a center frequency of a formant representing the formant position, a formant bandwidth, and a formant gain and reads out the band characteristics waveform at a readout interval, derived from the bandwidth wn, from a band characteristics waveform storage unit to effect expansion along the time axis.
    Type: Grant
    Filed: June 7, 2004
    Date of Patent: September 29, 2009
    Assignee: Sony Corporation
    Inventor: Nobuhide Yamazaki
  • Publication number: 20090171666
    Abstract: An interpolation device (4) includes a band extraction high-pass filter (11) for extracting a frequency component of a predetermined lower limit frequency or above from reproduction data obtained by digitizing an audio waveform signal; a multiplier (13) for frequency-shifting the frequency component extracted by the band extraction high-pass filter (11); lower side wave band suppression high-pass filter (14) suppressing the frequency component of the lower side wave band in the frequency component subjected to frequency shift by the multiplier (13); and an adder (17) for adding the frequency component after suppression by the lower side wave band suppression high-pass filter (14). It is possible to reduce the processing load.
    Type: Application
    Filed: November 29, 2006
    Publication date: July 2, 2009
    Applicant: Kabushiki Kaisha Kenwood
    Inventor: Hideki Ohtsu
  • Patent number: 7552052
    Abstract: A plurality of voice segments, each including one or more phonemes are acquired in a time-serial manner, in correspondence with desired singing or speaking words. As necessary, a boundary is designated between start and end points of a vowel phoneme included in any one of the acquired voice segments. Voice is synthesized for a region of the vowel phoneme that precedes the designated boundary vowel phoneme, or a region of the vowel phoneme that succeeds the designated boundary in the vowel phoneme. By synthesizing a voice for the region preceding the designated boundary, it is possible to synthesize a voice imitative of a vowel sound that is uttered by a person and then stopped to sound with his or her mouth kept opened. Further, by synthesizing a voice for the region succeeding the designated boundary, it is possible to synthesize a voice imitative of a vowel sound that is started to sound with the mouth opened.
    Type: Grant
    Filed: July 13, 2005
    Date of Patent: June 23, 2009
    Assignee: Yamaha Corporation
    Inventor: Hideki Kemmochi
  • Patent number: 7529672
    Abstract: A method of synthesizing a speech signal by providing a first speech unit signal having an end interval and a second speech unit signal having a front interval, wherein at least some of the periods of the end interval are appended in inverted order at the end of the first speech unit signal in order to provide a fade-out interval, and at least some of the periods of the front interval are appended in inverted order at the beginning of the second speech unit signal to provide a fade-in interval. An overlap and add operation is performed on the end and fade-in intervals and the fade-out and front intervals.
    Type: Grant
    Filed: August 8, 2003
    Date of Patent: May 5, 2009
    Assignee: Koninklijke Philips Electronics N.V.
    Inventor: Ercan Ferit Gigi
  • Patent number: 7529673
    Abstract: A method for use by a speech decoder in handling bad frames received over a communications channel a method in which the effects of bad frames are concealed by replacing the values of the spectral parameters of the bad frames (a bad frame being either a corrupted frame or a lost frame) with values based on an at least partly adaptive mean of recently received good frames, but in case of a corrupted frame (as opposed to a lost frame), using the bad frame itself if the bad frame meets a predetermined criterion. The aim of concealment is to find the most suitable parameters for the bad frame so that subjective quality of the synthesized speech is as high as possible.
    Type: Grant
    Filed: April 10, 2006
    Date of Patent: May 5, 2009
    Assignee: Nokia Corporation
    Inventors: Jari Mäkinen, Hannu Mikkola, Janne Vainio, Jani Rotola-Pukkila
  • Patent number: 7519535
    Abstract: A voice decoder configured to receive a sequence of frames, each of the frames having voice parameters. The voice decoder includes a speech generator that generates speech from the voice parameters. A frame erasure concealment module is configured to reconstruct the voice parameters for a frame erasure in the sequence of frames from the voice parameters in one of the previous frames and the voice parameters in one of the subsequent frames.
    Type: Grant
    Filed: January 31, 2005
    Date of Patent: April 14, 2009
    Assignee: QUALCOMM Incorporated
    Inventor: Serafin Diaz Spindola
  • Publication number: 20090070117
    Abstract: According to an aspect of an embodiment, a method for interpolating a partial loss of an audio signal including a sound signal component and a background noise component in transmission thereof, the method comprising the steps of: calculating frequency characteristic of the background noise in the audio signal; extracting the sound signal component from the audio signal; generating pseudo noise by applying the frequency characteristic of the background noise included in the audio signal to white noise; and generating an interpolation signal by combining the pseudo noise with the extracted sound signal component included in the audio signal to supersede the partial loss of the audio signal.
    Type: Application
    Filed: September 5, 2008
    Publication date: March 12, 2009
    Applicant: FUJITSU LIMITED
    Inventor: Kaori Endo
  • Patent number: 7478047
    Abstract: A system and method for controlling a synthetic character using a control system displays the character engaged in an activity, receiving a first input from a user, determines whether the input is relevant to the activity, if the input is relevant to the activity, and shows the character react to the input, the character being highly expressive and highly reactive. A system and method for displaying a synthetic character provides speech data, creates modified speech data by modifying at least one of the pitch or duration of at least a portion of the speech data and generates modified speech sounds associated with the character using the modified speech data.
    Type: Grant
    Filed: October 29, 2001
    Date of Patent: January 13, 2009
    Assignee: Zoesis, Inc.
    Inventors: A. Bryan Loyall, Joseph Bates, W. Scott Neal Reilly, Mark Russell Leone
  • Patent number: 7428492
    Abstract: The distance between the first two pitch marks of a voiced portion of speech data to be processed is calculated. The difference between the adjacent inter-pitch-mark distances is calculated. The respective calculation results are stored and managed in a file.
    Type: Grant
    Filed: February 2, 2006
    Date of Patent: September 23, 2008
    Assignee: Canon Kabushiki Kaisha
    Inventor: Masayuki Yamada
  • Publication number: 20080215330
    Abstract: A method of modifying an audio signal comprises the steps of analyzing the input audio signal (x) so as to produce a set of filter parameters (p) and a residual signal (r), modifying the set of filter parameters (p) so as to produce a modified set of filter parameters (p?), and synthesizing an output audio signal (y) using the modified set of filter parameters (p?) and the residual signal (r). The set of filter parameters (p) comprises poles (?A) and coefficients (a; c). The step of modifying the filter parameters (p) involves interpolating lattice filter reflection coefficients (c) so as to scale the spectral envelope of the audio signal.
    Type: Application
    Filed: July 18, 2006
    Publication date: September 4, 2008
    Applicant: KONINKLIJKE PHILIPS ELECTRONICS, N.V.
    Inventors: Aki Sakari Harma, Albertus Cornelis Den Brinker
  • Patent number: 7418388
    Abstract: A method and a system of producing a synthesized voice is provided. A voice sound waveform is provided at a voice sampling frequency based on pronunciation informations. A voice-less sound waveform is produced at a voice-less sampling frequency based on the pronunciation informations. The voice sampling frequency is converted into an output sampling frequency to produce a frequency-converted voice sound waveform with the output sampling frequency, wherein each of the voice sampling frequency and the voice-less sampling frequency is independent from the output sampling frequency. The voice-less sampling frequency is converted into the output sampling frequency to produce a frequency-converted voice-less sound waveform with the output sampling frequency.
    Type: Grant
    Filed: September 22, 2006
    Date of Patent: August 26, 2008
    Assignee: NEC Corporation
    Inventor: Reishi Kondo
  • Publication number: 20080195392
    Abstract: A bandwidth extension system extends the bandwidth of an acoustic signal. By shifting a portion of the signal by a frequency value, the system generates an upper bandwidth extension signal. An extended bandwidth acoustic signal may be generated from the acoustic signal, the upper bandwidth extension signal, and/or a lower bandwidth extension signal.
    Type: Application
    Filed: January 17, 2008
    Publication date: August 14, 2008
    Inventors: Bernd Iser, Gerhard Nussle, Gerhard Uwe Schmidt
  • Patent number: 7409347
    Abstract: Portions from segment boundary regions of a plurality of speech segments are extracted. Each segment boundary region is based on a corresponding initial unit boundary. Feature vectors that represent the portions in a vector space are created. For each of a plurality of potential unit boundaries within each segment boundary region, an average discontinuity based on distances between the feature vectors is determined. For each segment, the potential unit boundary associated with a minimum average discontinuity is selected as a new unit boundary.
    Type: Grant
    Filed: October 23, 2003
    Date of Patent: August 5, 2008
    Assignee: Apple Inc.
    Inventor: Jerome R. Bellegarda
  • Patent number: 7395202
    Abstract: Upon receiving (101) a vocoded voice frame and detecting (102) that the received vocoded voice frame comprises an erased frame, one automatically replaces (103) the erased frame with a valid frame having at least one error condition. In a preferred approach this error condition is one that is known to cause a receiving target platform to invoke a corresponding erasure process with respect to the valid frame when received.
    Type: Grant
    Filed: June 9, 2005
    Date of Patent: July 1, 2008
    Assignee: Motorola, Inc.
    Inventors: Mark D. Hetherington, Michael J. Kirk, Lee M. Proctor, Zhongwei Zhuang
  • Publication number: 20080140409
    Abstract: 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: Application
    Filed: September 12, 2006
    Publication date: June 12, 2008
    Inventor: David A. Kapilow
  • Publication number: 20080126096
    Abstract: An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme.
    Type: Application
    Filed: October 31, 2007
    Publication date: May 29, 2008
    Applicant: Samsung Electronics Co., Ltd.
    Inventors: Eun-mi OH, Ki-hyun Choo, Ho-sang Sung, Chang-yong Son, Jung-hoe Kim, Kang-eun Lee
  • Publication number: 20080071541
    Abstract: An audio signal interpolation device comprises a spectral movement calculation unit which determines a spectral movement which is indicative of a difference in each of spectral components between a frequency spectrum of a current frame of an input audio signal and a frequency spectrum of a previous frame of the input audio signal stored in a spectrum storing unit. An interpolation band determination unit determines a frequency band to be interpolated by using the frequency spectrum of the current frame and the spectral movement. A spectrum interpolation unit performs interpolation of spectral components in the frequency band for the current frame by using either the frequency spectrum of the current frame or the frequency spectrum of the previous frame.
    Type: Application
    Filed: July 25, 2007
    Publication date: March 20, 2008
    Applicant: Fujitsu Limited
    Inventors: Masakiyo Tanaka, Masanao Suzuki, Miyuki Shirakawa, Takashi Makiuchi
  • Patent number: 7318034
    Abstract: A voice signal interpolation apparatus is provided which can restore original human voices from human voices in a compressed state while maintaining a high sound quality. When a voice signal representative of a voice to be interpolated is acquired by a voice data input unit 1, a pitch deriving unit 2 filters this voice signal to identify a pitch length from the filtering result. A pitch length fixing unit 3 makes the voice signal have a constant time length of a section corresponding to a unit pitch, and generates pitch waveform data. A sub-band dividing unit 4 converts the pitch waveform data into sub-band data representative of a spectrum. A plurality of sub-band data pieces are averaged by an averaging unit 5 and thereafter a sub-band synthesizing unit 6 converts the sub-band data pieces into a signal representative of a waveform of the voice by a sub-band synthesizing unit 6.
    Type: Grant
    Filed: May 28, 2003
    Date of Patent: January 8, 2008
    Assignee: Kabushiki Kaisha Kenwood
    Inventor: Yasushi Sato
  • Patent number: 7307981
    Abstract: An apparatus for converting voice packets transmitted/received through a network includes a first transcoder for performing at least one of bit-unpacking and unquantization on an encoded packet at a first encoder, namely transmitting party, to obtain an LSP (Line Spectrum Pair) parameter of the first encoder, and converting and unquantizing the LSP parameter to an LSP parameter of a second encoder, namely receiving party, to do bit-packing. A second transcoder performs at least one of bit-unpacking and unquantization on an encoded packet at the second encoder, namely transmitting party, to obtain an LSP parameter of the second encoder, and converts and unquantizes the LSP parameter to an LSP parameter of the first encoder, namely receiving party, to do bit-packing.
    Type: Grant
    Filed: September 19, 2002
    Date of Patent: December 11, 2007
    Assignee: LG Electronics Inc.
    Inventors: Yong Soo Choi, Dae Hee Youn, Kyung Tae Kim
  • Patent number: 7283961
    Abstract: There is disclosed a speech processing device in which prediction taps for finding prediction values of the speech of high sound quality are extracted from the synthesized sound obtained on affording linear prediction coefficients and residual signals, generated from a preset code, to a speech synthesis filter, speech of high sound quality being higher in sound quality than the synthesized sound, and in which the prediction taps are used along with preset tap coefficients to perform preset predictive calculations to find the prediction values of the speech of high sound quality. The speech of high sound quality is higher in sound quality than the synthesized sound.
    Type: Grant
    Filed: August 3, 2001
    Date of Patent: October 16, 2007
    Assignee: Sony Corporation
    Inventors: Tetsujiro Kondo, Tsutomu Watanabe, Masaaki Hattori, Hiroto Kimura, Yasuhiro Fujimori
  • Patent number: 7280968
    Abstract: A method for digitally generating speech with improved prosodic characteristics can include receiving a speech input, determining at least one prosodic characteristic contained within the speech input, and generating a speech output including the prosodic characteristic within the speech output.
    Type: Grant
    Filed: March 25, 2003
    Date of Patent: October 9, 2007
    Assignee: International Business Machines Corporation
    Inventor: Oscar J. Blass
  • Patent number: 7263080
    Abstract: An integrated circuit for streaming media over wireless networks is disclosed. The integrated circuit includes a media module that is designed to process media data. When non-media data is in, switching means is provided to avoid the non-media data being processed in the media module. One of important features in the integrated circuit is the underlying designs that are capable of facilitating wireless communication in different wireless networks. In one embodiment, a baseband processor is provided to facilitate wireless communications in more than one standard. The baseband processor is uniquely designed to facilitate wireless communications in a Wi-Fi network as well as a WiMAX network. As a result, same chips may be used to stream media data across different wireless networks.
    Type: Grant
    Filed: April 15, 2006
    Date of Patent: August 28, 2007
    Assignee: RDW, Inc.
    Inventors: Robin Yubin Zhu, Chung-Hsing Chang, Ted Hsiung
  • Patent number: 7249020
    Abstract: A method and a system of producing a synthesized voice is provided. A voice sound waveform is provided at a voice sampling frequency based on pronunciation informations. A voice-less sound waveform is produced at a voice-less sampling frequency based on the pronunciation informations. The voice sampling frequency is converted into an output sampling frequency to produce a frequency-converted voice sound waveform with the output sampling frequency, wherein each of the voice sampling frequency and the voice-less sampling frequency is independent from the output sampling frequency. The voice-less sampling frequency is converted into the output sampling frequency to produce a frequency-converted voice-less sound waveform with the output sampling frequency.
    Type: Grant
    Filed: April 18, 2002
    Date of Patent: July 24, 2007
    Assignee: NEC Corporation
    Inventor: Reishi Kondo
  • Patent number: 7224853
    Abstract: A set of known data samples are identified and an approximation of an original function from which the known data samples were obtained is created. The approximation function is then resampled to obtain desired values that are not contained in the set of known data samples.
    Type: Grant
    Filed: May 29, 2002
    Date of Patent: May 29, 2007
    Assignee: Microsoft Corporation
    Inventor: Shankar Moni
  • Patent number: 7177812
    Abstract: A method for conversion of input audio frequency data, at an input sample frequency, to output audio frequency data, at an output sample frequency. The input data is subjected to expansion to produce expanded data at an output sample frequency. The expanded data is interpolated to produce output data. In one embodiment of the invention the interpolation is effected by a process that also filters the output data. In another embodiment, the input data is sampled by an integer factor to produce expanded data, the expanded data is then interpolated to produce the output data. Also disclosed is a method of transition of a signal output, at one frequency, to a signal output at another frequency. The signal output at said one frequency is faded out over a period, and the signal output at said other frequency is faded in over that period. Both signal outputs are combined to produce the signal output over said period. Apparatus for effecting the methods is also disclosed.
    Type: Grant
    Filed: June 23, 2000
    Date of Patent: February 13, 2007
    Assignee: STMicroelectronics Asia Pacific PTE Ltd
    Inventors: Mohammed Javed Absar, Sapna George, Antonio Mario Alvarez-Tinoco
  • Patent number: 7143032
    Abstract: A method and system are provided for removing discontinuities associated with synthesizing a corrupted frame output from a decoder including one or more predictive filters. The corrupted frame is representative of one segment of a decoded signal. The method comprises copying a first number of stored samples of the decoded signal in accordance with a time lag and a scaling factor, and calculating a first number of ringing samples output from at least one of the filters.
    Type: Grant
    Filed: June 28, 2002
    Date of Patent: November 28, 2006
    Assignee: Broadcom Corporation
    Inventor: Juin-Hwey Chen