Patents by Inventor Hong-Goo Kang
Hong-Goo Kang has filed for patents to protect the following inventions. This listing includes patent applications that are pending as well as patents that have already been granted by the United States Patent and Trademark Office (USPTO).
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Publication number: 20120123770Abstract: Disclosed is a method of improving a sound quality, including: receiving a transmission signal of a first user equipment; removing noise in the transmission signal using noise information of the first user equipment side; performing speech reinforcement with respect to the noise removed transmission signal using noise information of a second user equipment side; and transmitting the speech reinforced transmission signal to the second user equipment.Type: ApplicationFiled: September 12, 2011Publication date: May 17, 2012Applicant: Industry-Academic Cooperation Foundation, Yonsei UniversityInventors: Hong Goo Kang, Min Seok Choi, Ho Seon Shin
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Patent number: 8055499Abstract: The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.Type: GrantFiled: October 29, 2010Date of Patent: November 8, 2011Assignee: Electronics and Telecommunications Research InstituteInventors: Ho-Sang Sung, Dae-Hwan Hwang, Dae-Hee Youn, Hong-Goo Kang, Young-Cheol Park, Ki-Seung Lee, Sung-Kyo Jung, Kyung-Tae Kim
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Patent number: 7991621Abstract: An apparatus for processing an encoded signal and method thereof are disclosed, by which an audio signal can be compressed and reconstructed in higher efficiency. An audio signal processing method includes the steps of identifying whether a type of an audio signal is a music using first type information, if the type of the audio signal is not the music signal, identifying whether the type of the audio signal is a speech signal or a mixed signal using second type information, and if the type of the audio signal is determined as either the speech signal or the mixed signal, reconstructing the audio signal according to a coding scheme applied per frame using coding identification information. If the type of the audio signal is the music signal, the first type information is received only. If the type of the audio signal is the speech signal or the mixed signal, both of the first type information and the second type information are received.Type: GrantFiled: July 2, 2009Date of Patent: August 2, 2011Assignees: LG Electronics Inc., Industry-Academic Cooperation Foundation, Yonsei UniversityInventors: Hyen O Oh, Jeong Ook Song, Chang Heon Lee, Yang Won Jung, Hong Goo Kang
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Patent number: 7979272Abstract: The present invention provides a frame erasure concealment device and method that is based on reestimating gain parameters for a code excited linear prediction (CELP) coder. During operation, when a frame in a stream of received data is detected as being erased, the coding parameters, especially an adaptive codebook gain gp and a fixed codebook gain gc, of the erased and subsequent frames can be reestimated by a gain matching procedure. By using this technique with the IS-641 speech coder, it has been found that the present invention improves the speech quality under various channel conditions, compared with a conventional extrapolation-based concealment algorithm.Type: GrantFiled: October 12, 2007Date of Patent: July 12, 2011Assignee: AT&T Intellectual Property II, L.P.Inventors: Hong-Goo Kang, Hong Kook Kim
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Publication number: 20110153335Abstract: An audio signal processing method is disclosed. The audio signal processing method includes receiving a residual and long term prediction information, performing inverse frequency mapping with respect to the residual to generate a synthesized residual, and performing long term synthesis based on the synthesized residual and the long term prediction information to generate a synthesized audio signal of a current frame, wherein the long term prediction information comprises a final prediction gain and a final pitch lag, the final pitch lag has a range starting with 0, and the long term synthesis is performed based on a synthesized audio signal of a frame comprising a preceding frame.Type: ApplicationFiled: May 25, 2009Publication date: June 23, 2011Inventors: Hyen-O Oh, Chang Heon Lee, Jeongook Song, Yang Won Jung, Hong Goo Kang
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Publication number: 20110075855Abstract: A method for processing an audio signal is disclosed. The method for processing an audio signal includes frequency-transforming an audio signal to generate a frequency-spectrum, deciding a weighting per band corresponding to energy per band using the frequency spectrum, receiving a masking threshold based on a psychoacoustic model, applying the weighting to the masking threshold to generate a modified masking threshold, and quantizing the audio signal using the modified masking threshold.Type: ApplicationFiled: May 25, 2009Publication date: March 31, 2011Inventors: Hyen-O Oh, Chang Heon Lee, Jeongook Song, Yang Won Jung, Hong Goo Kang
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Publication number: 20110057818Abstract: Encoding and decoding apparatuses and encoding and decoding methods are provided. The decoding method includes extracting a plurality of encoded signals from an input bitstream, determining which of a plurality of decoding methods is to be used to decode each of the encoded signals, decoding the encoded signals using the determined decoding methods, and synthesizing the decoded signals. Accordingly, it is possible to encode signals having different characteristics at an optimum bitrate by classifying the signals into one or more classes according to the characteristics of the signals and encoding each of the signals using an encoding unit that can best serve the class where a corresponding signal belongs. In addition, it is possible to efficiently encode various signals including audio and speech signals.Type: ApplicationFiled: January 18, 2007Publication date: March 10, 2011Applicant: LG ELECTRONICS, INC.Inventors: Yang Won Jung, Hyen-O Oh, Hyo Jin Kim, Seung Jong Choi, Dong Geum Lee, Hong Goo Kang, Jae Seong Lee
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Publication number: 20110040557Abstract: The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.Type: ApplicationFiled: October 29, 2010Publication date: February 17, 2011Applicant: Electronics and Telecommunications Research InstituteInventors: Ho-Sang SUNG, Dae-Hwan Hwang, Dae-Hee Youn, Hong-Goo Kang, Young-Cheol Park, Ki-Seung Lee, Sung-Kyo Jung, Kyung-Tae Kim
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Publication number: 20110029317Abstract: Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder.Type: ApplicationFiled: July 30, 2010Publication date: February 3, 2011Applicant: BROADCOM CORPORATIONInventors: Juin-Hwey Chen, Hong-goo Kang, Robert W. Zopf, Jes Thyssen
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Publication number: 20110029304Abstract: A hybrid instantaneous/differential encoding technique is described herein that may be used to reduce the bit rate required to encode a pitch period associated with a segment of a speech signal in a manner that will result in relatively little or no degradation of a decoded speech signal generated using the encoded pitch period. The hybrid instantaneous/differential encoding technique is advantageously applicable to any speech codec that encodes a pitch period associated with a segment of a speech signal.Type: ApplicationFiled: July 30, 2010Publication date: February 3, 2011Applicant: BROADCOM CORPORATIONInventors: Juin-Hwey Chen, Hong-Goo Kang
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Patent number: 7860711Abstract: The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.Type: GrantFiled: February 22, 2008Date of Patent: December 28, 2010Assignee: Electronics and Telecommunications Research InstituteInventors: Ho-Sang Sung, Dae-Hwan Hwang, Dae-Hee Youn, Hong-Goo Kang, Young-Cheol Park, Ki-Seung Lee, Sung-Kyo Jung, Kyung-Tae Kim
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Publication number: 20100312551Abstract: Disclosed is a method of processing a signal, which includes receiving at least one of a first signal and a second signal, receiving mode information, and coding the at least one of the first signal and the second signal using at least one of a first coding scheme and a second coding scheme according to the mode information, wherein the mode information is information for indicating that a prescribed mode corresponds to which one of at least three modes.Type: ApplicationFiled: October 15, 2008Publication date: December 9, 2010Applicants: LG Electronics Inc., Industry-Academic Cooperation Foundation, Yonsei UniversityInventors: Hyen-O Oh, Hong Goo Kang, Chang Heon Lee, Sang Wook Shin, Yang Won Jung
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Publication number: 20100312567Abstract: Disclosed is a method of processing a signal, which includes receiving at least one of a first signal and a second signal, receiving mode information, and coding the at least one of the first signal and the second signal using at least one of a first coding scheme and a second coding scheme according to the mode information, wherein the mode information is information for indicating that a prescribed mode corresponds to which one of at least three modes.Type: ApplicationFiled: October 15, 2008Publication date: December 9, 2010Applicants: Industry-Academic Cooperation Foundation, Yonsei University, LG ELECTRONICS INC.Inventors: Hyen-O Oh, Hong Goo Kang, Chang Heon Lee, Sang Wook Shin, Yang Won Jung
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Patent number: 7711556Abstract: Methods and systems for filtering synthesized or reconstructed speech are implemented. A filter based on a set of linear predictive coding (LPC) coefficients is constructed by transforming the LPC coefficients to the pseudo-cepstrum, a domain existing between LPC domain and the line spectral frequency (LSF) domain. The resulting filter can emphasize spectral frequencies associated with various formants, or spectral peaks, of an inverse transfer function relating to the LPC coefficients, and can de-emphasize spectral frequencies associated with various spectral minima, or spectral valleys, of the inverse transfer function relating to the LPC coefficients.Type: GrantFiled: August 1, 2007Date of Patent: May 4, 2010Assignee: AT&T Intellectual Property II, L.P.Inventors: Hong-Goo Kang, Hong Kook Kim
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Publication number: 20100082335Abstract: 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: ApplicationFiled: December 4, 2009Publication date: April 1, 2010Applicant: Electronics and Telecommunications Research InstituteInventors: Ho-Sang SUNG, Dae-Hwan HWANG, Dae-Hee YOUN, Hong-Goo KANG, Young-Cheol PARK, Ki-Seung LEE, Sung-Kyo JUNG, Kyung-Tae KIM
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Publication number: 20100070284Abstract: An apparatus for processing an encoded signal and method thereof are disclosed, by which an audio signal can be compressed and reconstructed in higher efficiency. An audio signal processing method includes the steps of identifying whether a type of an audio signal is a music using first type information, if the type of the audio signal is not the music signal, identifying whether the type of the audio signal is a speech signal or a mixed signal using second type information, and if the type of the audio signal is determined as either the speech signal or the mixed signal, reconstructing the audio signal according to a coding scheme applied per frame using coding identification information. If the type of the audio signal is the music signal, the first type information is received only. If the type of the audio signal is the speech signal or the mixed signal, both of the first type information and the second type information are received.Type: ApplicationFiled: July 2, 2009Publication date: March 18, 2010Applicants: LG Electronics Inc., INDUSTRY-ACADEMIC COOPERATION FORUNDATION, YONSEI UNIVERSITYInventors: Hyen O. OH, Jeong Ook Song, Chang Heon Lee, Yang Won Jung, Hong Goo Kang
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Patent number: 7649856Abstract: 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: GrantFiled: December 18, 2003Date of Patent: January 19, 2010Assignee: Electronics and Telecommunications Research InstituteInventors: Ho-Sang Sung, Dae-Hwan Hwang, Dae-Hee Youn, Hong-Goo Kang, Young-Cheol Park, Ki-Seung Lee, Sung-Kyo Jung, Kyung-Tae Kim
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Publication number: 20090281812Abstract: Encoding and decoding apparatuses and encoding and decoding methods are provided. The decoding method includes extracting a plurality of encoded signals and division information of the encoded signals from an input bitstream, determining which of a plurality of decoding methods is to be used to decode each of the encoded signals, decoding the encoded signals using the determined decoding methods, and synthesizing the decoded signals with reference to the division information. Accordingly, it is possible to encode signals having different characteristics at an optimum bitrate by classifying the signals into one or more classes according to the characteristics of the signals and encoding each of the signals using an encoding unit that can best serve the class where a corresponding signal belongs. In addition, it is possible to efficiently encode various signals including audio and speech signals.Type: ApplicationFiled: January 18, 2007Publication date: November 12, 2009Applicant: LG ELECTRONICS INC.Inventors: Yang Won Jung, Hyen-O Oh, Hyo Jin Kim, Seung Jong Choi, Dong Geum Lee, Hong Goo Kang, Jae Seong Lee
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Publication number: 20090222261Abstract: Encoding and decoding apparatuses and encoding and decoding methods are provided. The decoding method includes extracting a plurality of encoded signals from an input bitstream, determining which of a plurality of decoding methods is to be used to decode each of the encoded signals, decoding the encoded signals using the determined decoding methods, synthesizing the decoded signals, and restoring an original signal by performing a post-processing operation on the single signal. Accordingly, it is possible to encode signals having different characteristics at an optimum bitrate by classifying the signals into one or more classes according to the characteristics of the signals and encoding each of the signals using an encoding unit that can best serve the class where a corresponding signal belongs. In addition, it is possible to efficiently encode various signals including audio and speech signals.Type: ApplicationFiled: January 18, 2007Publication date: September 3, 2009Applicant: LG ELECTRONICS, INC.Inventors: Yang Won Jung, Hyun O Oh, Hyo Jin Kim, Seung Jong Choi, Dong Geum Lee, Hong Goo Kang, Jae Seong Lee, Young Cheol Park
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Patent number: 7472056Abstract: A transcoder for use between speech codecs using different Code-Excited Linear Prediction (CELP) type and a method therefor are disclosed. The transcoder includes a decoding unit of an input CELP codec, a transcoding filter, a transcoding filter design unit, and an encoding unit of an output CELP codec. By substituting a post-filter and a perceptual weighting filter of a prior art with one transcoding filter, the calculation amount of the transcoder is reduced, and speech quality decoded at a receiving end is improved.Type: GrantFiled: December 30, 2003Date of Patent: December 30, 2008Assignee: Electronics and Telecommunications Research InstituteInventors: Jongmo Sung, Hyun Woo Kim, Do Young Kim, Jin Kyu Choi, Sung Wan Yoon, Hong Goo Kang, Ki Seung Lee, Dae Hee Youn