Patents by Inventor Kazuhito Koishida
Kazuhito Koishida 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: 20080040105Abstract: Techniques and tools related to coding and decoding of audio information are described. For example, redundant coded information for decoding a current frame includes signal history information associated with only a portion of a previous frame. As another example, redundant coded information for decoding a coded unit includes parameters for a codebook stage to be used in decoding the current coded unit only if the previous coded unit is not available. As yet another example, coded audio units each include a field indicating whether the coded unit includes main encoded information representing a segment of an audio signal, and whether the coded unit includes redundant coded information for use in decoding main encoded information.Type: ApplicationFiled: October 9, 2007Publication date: February 14, 2008Applicant: Microsoft CorporationInventors: Tian Wang, Kazuhito Koishida, Hosam Khalil, Xiaoqin Sun, Wei-Ge Chen
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Publication number: 20080040121Abstract: Techniques and tools related to coding and decoding of audio information are described. For example, redundant coded information for decoding a current frame includes signal history information associated with only a portion of a previous frame. As another example, redundant coded information for decoding a coded unit includes parameters for a codebook stage to be used in decoding the current coded unit only if the previous coded unit is not available. As yet another example, coded audio units each include a field indicating whether the coded unit includes main encoded information representing a segment of an audio signal, and whether the coded unit includes redundant coded information for use in decoding main encoded information.Type: ApplicationFiled: October 9, 2007Publication date: February 14, 2008Applicant: Microsoft CorporationInventors: Tian Wang, Kazuhito Koishida, Hosam Khalil, Xiaoqin Sun, Wei-Ge Chen
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Patent number: 7315815Abstract: An enhanced low-bit rate parametric voice coder that groups a number of frames from an underlying frame-based vocoder, such as MELP, into a superframe structure. Parameters are extracted from the group of underlying frames and quantized into the superframe which allows the bit rate of the underlying coding to be reduced without increasing the distortion. The speech data coded in the superframe structure can then be directly synthesized to speech or may be transcoded to a format so that an underlying frame-based vocoder performs the synthesis. The superframe structure includes additional error detection and correction data to reduce the distortion caused by the communication of bit errors.Type: GrantFiled: September 22, 1999Date of Patent: January 1, 2008Assignee: Microsoft CorporationInventors: Allen Gersho, Vladimir Cuperman, Tian Wang, Kazuhito Koishida
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Patent number: 7286982Abstract: An enhanced low-bit rate parametric voice coder that groups a number of frames from an underlying frame-based vocoder, such as MELP, into a superframe structure. Parameters are extracted from the group of underlying frames and quantized into the superframe which allows the bit rate of the underlying coding to be reduced without increasing the distortion. The speech data coded in the superframe structure can then be directly synthesized to speech or may be transcoded to a format so that an underlying frame-based vocoder performs the synthesis. The superframe structure includes additional error detection and correction data to reduce the distortion caused by the communication of bit errors.Type: GrantFiled: July 20, 2004Date of Patent: October 23, 2007Assignee: Microsoft CorporationInventors: Allen Gersho, Vladimir Cuperman, Tian Wang, Kazuhito Koishida
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Patent number: 7280960Abstract: Techniques and tools related to coding and decoding of audio information are described. For example, redundant coded information for decoding a current frame includes signal history information associated with only a portion of a previous frame. As another example, redundant coded information for decoding a coded unit includes parameters for a codebook stage to be used in decoding the current coded unit only if the previous coded unit is not available. As yet another example, coded audio units each include a field indicating whether the coded unit includes main encoded information representing a segment of an audio signal, and whether the coded unit includes redundant coded information for use in decoding main encoded information.Type: GrantFiled: August 4, 2005Date of Patent: October 9, 2007Assignee: Microsoft CorporationInventors: Tian Wang, Kazuhito Koishida, Hosam A. Khalil, Xiaoqin Sun, Wei-Ge Chen
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Publication number: 20070174063Abstract: An audio encoder performs frequency extension coding that comprises determining one or more shape parameters using a displacement vector that corresponds to a displacement of an even number (e.g., an even number of sub-bands between a sub-band in a baseband frequency range and a sub-band in an extended-band frequency range). The shape parameters can be determined on a per-audio-block basis. Restricting a displacement to an even number (in frequency extension coding or in other signal modulation schemes) can improve the quality of reconstructed audio. An audio encoder also can perform frequency extension coding that comprises determining one or more scale parameters at one or more audio blocks, and determining one or more anchor points for interpolating the one or more scale parameters.Type: ApplicationFiled: January 20, 2006Publication date: July 26, 2007Applicant: Microsoft CorporationInventors: Sanjeev Mehrotra, Wei-Ge Chen, Kazuhito Koishida, Chao He
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Patent number: 7177804Abstract: Techniques and tools related to coding and decoding of audio information are described. For example, redundant coded information for decoding a current frame includes signal history information associated with only a portion of a previous frame. As another example, redundant coded information for decoding a coded unit includes parameters for a codebook stage to be used in decoding the current coded unit only if the previous coded unit is not available. As yet another example, coded audio units each include a field indicating whether the coded unit includes main encoded information representing a segment of an audio signal, and whether the coded unit includes redundant coded information for use in decoding main encoded information.Type: GrantFiled: May 31, 2005Date of Patent: February 13, 2007Assignee: Microsoft CorporationInventors: Tian Wang, Kazuhito Koishida, Hosam A. Khalil, Xiaoqin Sun, Wei-Ge Chen
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Publication number: 20070016414Abstract: Coding of spectral data by representing certain portions of the spectral data as a scaled version of a code-vector, where the code-vector is chosen from either a fixed predetermined codebook or a codebook taken from a baseband. Various optional features are described for modifying the code-vectors in the codebook according to some rules which allow the code-vector to better represent the data they are modeling. The code-vector modification comprises a linear or non-linear transform of one or more code-vectors, such as, by exponentiation, negation, reversing, or combining elements from plural code-vectors.Type: ApplicationFiled: July 15, 2005Publication date: January 18, 2007Applicant: Microsoft CorporationInventors: Sanjeev Mehrotra, Wei-Ge Chen, Kazuhito Koishida
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Publication number: 20060271373Abstract: Techniques and tools related to delayed or lost coded audio information are described. For example, a concealment technique for one or more missing frames is selected based on one or more factors that include a classification of each of one or more available frames near the one or more missing frames. As another example, information from a concealment signal is used to produce substitute information that is relied on in decoding a subsequent frame. As yet another example, a data structure having nodes corresponding to received packet delays is used to determine a desired decoder packet delay value.Type: ApplicationFiled: May 31, 2005Publication date: November 30, 2006Applicant: Microsoft CorporationInventors: Hosam Khalil, Tian Wang, Kazuhito Koishida, Xiaoqin Sun, Wei-Ge Chen
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Publication number: 20060271355Abstract: Techniques and tools related to coding and decoding of audio information are described. For example, redundant coded information for decoding a current frame includes signal history information associated with only a portion of a previous frame. As another example, redundant coded information for decoding a coded unit includes parameters for a codebook stage to be used in decoding the current coded unit only if the previous coded unit is not available. As yet another example, coded audio units each include a field indicating whether the coded unit includes main encoded information representing a segment of an audio signal, and whether the coded unit includes redundant coded information for use in decoding main encoded information.Type: ApplicationFiled: May 31, 2005Publication date: November 30, 2006Applicant: Microsoft CorporationInventors: Tian Wang, Kazuhito Koishida, Hosam Khalil, Xiaoqin Sun, Wei-Ge Chen
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Publication number: 20060271359Abstract: Techniques and tools related to delayed or lost coded audio information are described. For example, a concealment technique for one or more missing frames is selected based on one or more factors that include a classification of each of one or more available frames near the one or more missing frames. As another example, information from a concealment signal is used to produce substitute information that is relied on in decoding a subsequent frame. As yet another example, a data structure having nodes corresponding to received packet delays is used to determine a desired decoder packet delay value.Type: ApplicationFiled: August 4, 2005Publication date: November 30, 2006Applicant: Microsoft CorporationInventors: Hosam Khalil, Tian Wang, Kazuhito Koishida, Xiaoqin Sun, Wei-Ge Chen
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Publication number: 20060271354Abstract: Techniques and tools are described for processing reconstructed audio signals. For example, a reconstructed audio signal is filtered in the time domain using filter coefficients that are calculated, at least in part, in the frequency domain. As another example, producing a set of filter coefficients for filtering a reconstructed audio signal includes clipping one or more peaks of a set of coefficient values. As yet another example, for a sub-band codec, in a frequency region near an intersection between two sub-bands, a reconstructed composite signal is enhanced.Type: ApplicationFiled: May 31, 2005Publication date: November 30, 2006Applicant: Microsoft CorporationInventors: Xiaoqin Sun, Tian Wang, Hosam Khalil, Kazuhito Koishida, Wei-Ge Chen
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Publication number: 20060271357Abstract: Techniques and tools related to coding and decoding of audio information are described. For example, redundant coded information for decoding a current frame includes signal history information associated with only a portion of a previous frame. As another example, redundant coded information for decoding a coded unit includes parameters for a codebook stage to be used in decoding the current coded unit only if the previous coded unit is not available. As yet another example, coded audio units each include a field indicating whether the coded unit includes main encoded information representing a segment of an audio signal, and whether the coded unit includes redundant coded information for use in decoding main encoded information.Type: ApplicationFiled: August 4, 2005Publication date: November 30, 2006Applicant: Microsoft CorporationInventors: Tian Wang, Kazuhito Koishida, Hosam Khalil, Xiaoqin Sun, Wei-Ge Chen
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Publication number: 20050278172Abstract: A gain-constrained noise suppression for speech more precisely estimates noise, including during speech, to reduce musical noise artifacts introduced from noise suppression. The noise suppression operates by applying a spectral gain G(m, k) to each short-time spectrum value S(m, k) of a speech signal, where m is the frame number and k is the spectrum index. The spectrum values are grouped into frequency bins, and a noise characteristic estimated for each bin classified as a “noise bin.” An energy parameter is smoothed in both the time domain and the frequency domain to improve noise estimation per bin. The gain factors G(m, k) are calculated based on the current signal spectrum and the noise estimation, then smoothed before being applied to the signal spectral values S(m, k).Type: ApplicationFiled: June 15, 2004Publication date: December 15, 2005Applicant: Microsoft CorporationInventors: Kazuhito Koishida, Feng Zhuge, Hosam Khalil, Tian Wang, Wei-ge Chen
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Publication number: 20050228651Abstract: Various strategies for rate/quality control and loss resiliency in an audio codec are described. The various strategies can be used in combination or independently. For example, a real-time speech codec uses intra frame coding/decoding, adaptive multi-mode forward error correction [“FEC”], and rate/quality control techniques. Intra frames help a decoder recover quickly from packet losses, while compression efficiency is still emphasized with predicted frames. Various strategies for inserting intra frames and signaling intra/predicted frames are described. With the adaptive multi-mode FEC, an encoder adaptively selects between multiple modes to efficiently and quickly provide a level of FEC that takes into account the bandwidth currently available for FEC. The FEC information itself may be predictively encoded and decoded relative to primary encoded information. Various rate/quality and FEC control strategies allow additional adaptation to available bandwidth and network conditions.Type: ApplicationFiled: March 31, 2004Publication date: October 13, 2005Inventors: Tian Wang, Hosam Khalil, Kazuhito Koishida, Wei-Ge Chen, Mu Han
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Publication number: 20050075869Abstract: An enhanced_low-bit rate parametric voice coder that groups a number of frames from an underlying frame-based vocoder, such as MELP, into a superframe structure. Parameters are extracted from the group of underlying frames and quantized into the superframe which allows the bit rate of the underlying coding to be reduced without increasing the distortion. The speech data coded in the superframe structure can then be directly synthesized to speech or may be transcoded to a format so that an underlying frame-based vocoder performs the synthesis. The superframe structure includes additional error detection and correction data to reduce the distortion caused by the communication of bit errors.Type: ApplicationFiled: July 20, 2004Publication date: April 7, 2005Applicant: Microsoft CorporationInventors: Allen Gersho, Vladimir Cuperman, Tian Wang, Kazuhito Koishida
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Patent number: 6658383Abstract: The present invention provides a transform coding method efficient for music signals that is suitable for use in a hybrid codec, whereby a common Linear Predictive (LP) synthesis filter is employed for both speech and music signals. The LP synthesis filter switches between a speech excitation generator and a transform excitation generator, in accordance with the coding of a speech or music signal, respectively. For coding speech signals, the conventional CELP technique may be used, while a novel asymmetrical overlap-add transform technique is applied for coding music signals. In performing the common LP synthesis filtering, interpolation of the LP coefficients is conducted for signals in overlap-add operation regions. The invention enables smooth transitions when the decoder switches between speech and music decoding modes.Type: GrantFiled: June 26, 2001Date of Patent: December 2, 2003Assignee: Microsoft CorporationInventors: Kazuhito Koishida, Vladimir Cuperman, Amir H. Majidimehr, Allen Gersho
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Patent number: 6647366Abstract: A method and a system are provided for controlling the coding rates of a multimode coding system with respect to a sequence of input audio signal frames. The method eliminates or minimizes the overflow and underflow of a bit-stream buffer maintained by the coding system for temporarily recording bit-stream data prior to transmission or storage.Type: GrantFiled: December 28, 2001Date of Patent: November 11, 2003Assignee: Microsoft CorporationInventors: Tian Wang, Kazuhito Koishida, Vladimir Cuperman
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Publication number: 20030125932Abstract: A method and a system are provided for controlling the coding rates of a multimode coding system with respect to a sequence of input audio signal frames. The method eliminates or minimizes the overflow and underflow of a bit-stream buffer maintained by the coding system for temporarily recording bit-stream data prior to transmission or storage.Type: ApplicationFiled: December 28, 2001Publication date: July 3, 2003Applicant: Microsoft CorporationInventors: Tian Wang, Kazuhito Koishida, Vladimir Cuperman
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Publication number: 20030004711Abstract: The present invention provides a transform coding method efficient for music signals that is suitable for use in a hybrid codec, whereby a common Linear Predictive (LP) synthesis filter is employed for both speech and music signals. The LP synthesis filter switches between a speech excitation generator and a transform excitation generator, in accordance with the coding of a speech or music signal, respectively. For coding speech signals, the conventional CELP technique may be used, while a novel asymmetrical overlap-add transform technique is applied for coding music signals. In performing the common LP synthesis filtering, interpolation of the LP coefficients is conducted for signals in overlap-add operation regions. The invention enables smooth transitions when the decoder switches between speech and music decoding modes.Type: ApplicationFiled: June 26, 2001Publication date: January 2, 2003Applicant: Microsoft CorporationInventors: Kazuhito Koishida, Vladimir Cuperman, Amir H. Majidimehr, Allen Gersho