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    Synchronizing timing mismatch by data insertion
    1.
    发明公开
    Synchronizing timing mismatch by data insertion 审中-公开
    通过数据插入同步时序错误

    公开(公告)号：KR20120059651A

    公开(公告)日：2012-06-08

    申请号：KR20127012786

    申请日：2008-05-12

    Applicant: QUALCOMM INC

    Inventor： GARUDADRI HARINATH ,  MAJUMDAR SOMDEB ,  KASHEF ROUZBEH ,  GANAPATHY CHINNAPPA K

    IPC: H04L7/00 ,  H04L25/05

    CPC classification number: H04L7/0016 ,  G10L19/005 ,  H04L7/0029 ,  H04L7/005 ,  H04L25/05

    Abstract: 데이터가 하나의 디바이스에 의해 제공되는 레이트 및 데이터가 다른 디바이스에 의해 프로세싱되는 레이트는 상이할 수 있다. 예를 들어, 송신 디바이스는 송신 클럭에 따라 데이터를 송신할 수 있는 반면, 송신된 데이터를 수신하는 수신 디바이스는 수신 클럭에 따라 데이터를 프로세싱할 수 있다. 송신 및 수신 클럭들 사이의 타이밍 미스매치가 존재하는 경우, 수신 디바이스는 그것이 데이터를 프로세싱하는 것보다 더 빠르게 또는 더 느리게 데이터를 수신할 수 있다. 이러한 경우, 수신된 데이터의 프로세싱과 관련된 에러들이 존재할 수 있다. 이와 같은 타이밍 미스매치들을 처리하기 위해서, 수신 디바이스는 수신된 데이터로부터 데이터를 삭제하거나, 또는 수신된 데이터에 데이터를 삽입할 수 있다. 이러한 연산들과 관련하여, 수신 디바이스는 삽입 또는 삭제가 결과적인 출력 신호 상에서 미칠 수 있는 임의의 역효과를 완화시키는 방식으로 삽입 지점 또는 삭제 지점에서 또는 그 부근에서 수신된 데이터를 수정할 수 있다.

    Synchronizing timing mismatch by data deletion
    2.
    发明公开
    Synchronizing timing mismatch by data deletion 审中-公开
    通过数据删除同步时序错误

    公开(公告)号：KR20120059652A

    公开(公告)日：2012-06-08

    申请号：KR20127012790

    申请日：2008-05-12

    Applicant: QUALCOMM INC

    Inventor： GARUDADRI HARINATH ,  MAJUMDAR SOMDEB ,  KASHEF ROUZBEH ,  GANAPATHY CHINNAPPA K

    IPC: H04L7/00 ,  H04L25/05

    CPC classification number: H04L7/0016 ,  G10L19/005 ,  H04L7/0029 ,  H04L7/005 ,  H04L25/05

    Abstract: 데이터가 하나의 디바이스에 의해 제공되는 레이트 및 데이터가 다른 디바이스에 의해 프로세싱되는 레이트는 상이할 수 있다. 예를 들어, 송신 디바이스는 송신 클럭에 따라 데이터를 송신할 수 있는 반면, 송신된 데이터를 수신하는 수신 디바이스는 수신 클럭에 따라 데이터를 프로세싱할 수 있다. 송신 및 수신 클럭들 사이의 타이밍 미스매치가 존재하는 경우, 수신 디바이스는 그것이 데이터를 프로세싱하는 것보다 더 빠르게 또는 더 느리게 데이터를 수신할 수 있다. 이러한 경우, 수신된 데이터의 프로세싱과 관련된 에러들이 존재할 수 있다. 이와 같은 타이밍 미스매치들을 처리하기 위해서, 수신 디바이스는 수신된 데이터로부터 데이터를 삭제하거나, 또는 수신된 데이터에 데이터를 삽입할 수 있다. 이러한 연산들과 관련하여, 수신 디바이스는 삽입 또는 삭제가 결과적인 출력 신호 상에서 미칠 수 있는 임의의 역효과를 완화시키는 방식으로 삽입 지점 또는 삭제 지점에서 또는 그 부근에서 수신된 데이터를 수정할 수 있다.

    Transform design with scaled and non-scaled interface
    3.
    发明专利
    Transform design with scaled and non-scaled interface 有权
    变换设计与标准和非标准接口

    公开(公告)号：JP2012105273A

    公开(公告)日：2012-05-31

    申请号：JP2011243719

    申请日：2011-11-07

    Applicant: Qualcomm Inc ,  クゥアルコム・インコーポレイテッドQualcomm Incorporated

    Inventor： YURIY REZNIK ,  ALBERT SCOTT LUDWIG ,  CHUNG HYUKJUNE ,  GARUDADRI HARINATH ,  NAVEEN B SRINIVASAMASI ,  FORM SAGETON

    IPC: H03M7/30 ,  H04N1/41 ,  H04N7/30

    CPC classification number: G06F17/14 ,  G06F17/147 ,  H04N19/42 ,  H04N19/61

    Abstract: PROBLEM TO BE SOLVED: To efficiently perform transforms such as DCT and IDCT on a large quantity of data of various applications.SOLUTION: A full transform is a transform that implements the complete mathematical description of the transform. A scaled transform is a transform that operates on or provides scaled transform coefficients, which are scaled versions of full transform coefficients. The scaled transform may have lower computational complexity whereas the full transform may be simpler to use by applications. The full and scaled transforms may be for a 2D IDCT, which may be implemented in a separable manner with 1D IDCTs. The full and scaled transforms may also be for a 2D DCT, which may be implemented in a separable manner with ID DCTs. The 1D IDCTs and 1D DCTs may be implemented in a computationally efficient manner.

    Abstract translation: 要解决的问题：为了在各种应用的大量数据上有效地执行诸如DCT和IDCT之类的变换。 解决方案：全变换是实现变换的完整数学描述的变换。 缩放变换是对或者提供缩放变换系数的变换，其是全变换系数的缩放版本。 缩放变换可能具有较低的计算复杂度，而全变换可能更易于由应用使用。 完整和缩放的变换可以用于2D IDCT，其可以用1D IDCT以可分离的方式实现。 完整和缩放的变换也可以用于2D DCT，其可以用ID DCT以可分离的方式实现。 1D IDCT和1D DCT可以以计算有效的方式实现。 版权所有（C）2012，JPO＆INPIT

    Voice recognition system using implicit speaker adaption
    4.
    发明专利
    Voice recognition system using implicit speaker adaption 有权
    语音识别系统使用隐含的扬声器自适应

    公开(公告)号：JP2008077099A

    公开(公告)日：2008-04-03

    申请号：JP2007279235

    申请日：2007-10-26

    Applicant: Qualcomm Inc ,  クゥアルコム・インコーポレイテッドQualcomm Incorporated

    Inventor： MALAYATH NARENDRANATH ,  DEJACO ANDREW P ,  CHANG CHIENCHUNG ,  JALIL SUHAIL ,  BI NING ,  GARUDADRI HARINATH

    IPC: G10L15/06 ,  G10L15/10 ,  G10L15/12 ,  G10L15/14 ,  G10L15/16 ,  G10L17/04

    CPC classification number: G10L15/142 ,  G10L15/07 ,  G10L15/10 ,  G10L15/12 ,  G10L15/144 ,  G10L15/32

    Abstract: PROBLEM TO BE SOLVED: To provide a voice recognition method that excellently functions even before and during generation of a speaker dependent (SD) acoustic model to which unsupervised training is applied.
    SOLUTION: A voice recognition (VR) system is disclosed that utilizes at least one speaker independent (SI) acoustic model 230 or 232 in combination with at least one speaker dependent (SD) acoustic model 234 to provide a level of speech recognition performance that at least equals that of the purely SI acoustic model 230 or 232. The disclosed VR system continually uses unsupervised training to update the acoustic templates in the one or more acoustic models. The VR system then uses the updated SD acoustic model 234 in combination with the at least one SI acoustic model 230 or 232. Consequently, improved VR performance is obtained even during VR testing.
    COPYRIGHT: (C)2008,JPO&INPIT

    Abstract translation: 要解决的问题：提供一种即使在应用无监督训练的说话者依赖（SD）声学模型的生成之前和期间也能够出色地起作用的语音识别方法。 解决方案：公开了一种语音识别（VR）系统，其利用与至少一个说话者依赖（SD）声学模型234组合的至少一个扬声器独立（SI）声学模型230或232，以提供一定程度的语音识别 性能至少等于纯SI声学模型230或232的性能。所公开的VR系统连续地使用无监督训练来更新一个或多个声学模型中的声学模板。 VR系统然后使用更新的SD声学模型234与至少一个SI声学模型230或232组合。因此，即使在VR测试期间也获得了改进的VR性能。 版权所有（C）2008，JPO＆INPIT

    Content- and link-dependent coding adaptation for multimedia telephony
    5.
    发明专利
    Content- and link-dependent coding adaptation for multimedia telephony 审中-公开
    多媒体电话的内容和链接相关编码适应

    公开(公告)号：JP2012231467A

    公开(公告)日：2012-11-22

    申请号：JP2012099738

    申请日：2012-04-25

    Applicant: Qualcomm Inc ,  クゥアルコム・インコーポレイテッドQualcomm Incorporated

    Inventor： GARUDADRI HARINATH ,  CHANDRU SUNDARRAMAN ,  GAJINDER VIJ ,  NAVEEN B SRINIVASAMASI ,  FORM SAGETON

    IPC: H04L12/801 ,  H04M3/00 ,  H04L29/06 ,  H04N7/14 ,  H04W28/22

    CPC classification number: H04L65/4092 ,  H04L47/10 ,  H04L65/1059 ,  H04L65/602 ,  H04L65/80 ,  H04W72/08

    Abstract: PROBLEM TO BE SOLVED: To provide techniques that can facilitate multimedia telephony.SOLUTION: A method for communication of multimedia data comprises: determining a first level of throughput associated with multimedia data communication from a first access terminal to a network; determining a second level of throughput associated with multimedia data communication from the network to a second access terminal based on feedback from the second access terminal to the first access terminal via the network; determining a budget associated with communication of a video unit of the multimedia data; and coding the video unit of the multimedia data on the basis of the budget and the first and second levels of throughput.

    Abstract translation: 要解决的问题：提供可以促进多媒体电话的技术。 解决方案：一种用于多媒体数据通信的方法包括：确定与从第一接入终端到网络的多媒体数据通信相关联的吞吐量的第一级; 基于从所述第二接入终端经由所述网络向所述第一接入终端的反馈，确定与从所述网络到所述第二接入终端的多媒体数据通信相关联的吞吐量的第二级别; 确定与所述多媒体数据的视频单元的通信相关联的预算; 并根据预算和第一和第二吞吐量来编码多媒体数据的视频单元。 版权所有（C）2013，JPO＆INPIT

    Speech recognition system using technology for implicitly adapting to speaker
    7.
    发明专利
    Speech recognition system using technology for implicitly adapting to speaker 审中-公开
    使用技术的语音识别系统，用于明确适应扬声器

    公开(公告)号：JP2013152475A

    公开(公告)日：2013-08-08

    申请号：JP2013041687

    申请日：2013-03-04

    Applicant: Qualcomm Inc ,  クゥアルコム・インコーポレイテッドQualcomm Incorporated

    Inventor： MALAYATH NARENDRANATH ,  DEJACO ANDREW P ,  CHANG CHIENCHUNG ,  SUHAIL JALIL ,  BI NING ,  GARUDADRI HARINATH

    IPC: G10L15/06 ,  G10L15/32 ,  G10L15/065 ,  G10L15/10 ,  G10L15/12 ,  G10L15/14 ,  G10L15/16 ,  G10L17/04

    CPC classification number: G10L15/142 ,  G10L15/07 ,  G10L15/10 ,  G10L15/12 ,  G10L15/144 ,  G10L15/32

    Abstract: PROBLEM TO BE SOLVED: To attain a speech recognition method for successfully functioning even before generation and during the generation of an SD acoustic model depending on a speaker to whom unmanaged learning is applied.SOLUTION: In order to give a speech recognition level at least equal to pure SI acoustic models 230, 232 which do not depend on a speaker, at least one SI acoustic model 230, 232 and at least one SD acoustic model 234 which depends on the speaker are combined and used. Then, acoustic templates in at least one or more acoustic models are updated by continuously performing unmanaged learning. After that, the updated SD acoustic model 234 is combined with at least one SI acoustic model 230, 232. Thus, high speech recognition performance is obtained even while a speech recognition test is performed.

    Abstract translation: 要解决的问题：实现即使在生成之前和在SD声学模型的生成期间成功运行的语音识别方法，取决于应用非管理学习的说话者。解决方案：为了给予语音识别水平至少等于 对于不依赖于扬声器的纯SI声学模型230,232，组合并使用取决于扬声器的至少一个SI声学模型230,232和至少一个SD声学模型234。 然后，通过连续执行非管理学习来更新至少一个或多个声学模型中的声学模板。 之后，更新的SD声学模型234与至少一个SI声学模型230,232组合。因此，即使在执行语音识别测试时也获得高语音识别性能。

    Method and apparatus for signal processing using transform-domain log-companding
    8.
    发明专利
    Method and apparatus for signal processing using transform-domain log-companding 审中-公开
    使用变换域日志控制进行信号处理的方法和装置

    公开(公告)号：JP2013081229A

    公开(公告)日：2013-05-02

    申请号：JP2012268459

    申请日：2012-12-07

    Applicant: Qualcomm Inc ,  クゥアルコム・インコーポレイテッドQualcomm Incorporated

    Inventor： GARUDADRI HARINATH ,  SHUE YEN-LIANG ,  SOMDEB MAJUMDAR

    IPC: H03M7/50 ,  G10L19/02

    CPC classification number: H03M7/50 ,  G10L19/0204 ,  G10L19/0212 ,  G10L19/032

    Abstract: PROBLEM TO BE SOLVED: To provide a method and apparatus for audio signal processing by applying log companding on spectral domain or time domain representations of the audio signals to provide an encoded audio signal.SOLUTION: The encoded audio signal is decoded upon receipt. A frequency domain representation or time domain representation of the audio signal is computed by separating the audio signal into specific frequency bands, each having a coefficient. Log companding with different compression ratios is performed on each coefficient to provide an encoded signal. Upon receipt of the encoded signal, inverse log companding and time frequency or time scale reconstruction are performed to provide the audio signal.

    Abstract translation: 要解决的问题：提供一种用于音频信号处理的方法和装置，通过在音频信号的频域或时域表示上应用对数压扩扩展以提供经编码的音频信号。 解决方案：编码的音频信号在接收时被解码。 通过将音频信号分离成各自具有系数的特定频带来计算音频信号的频域表示或时域表示。 对每个系数执行具有不同压缩比的对数压扩比以提供编码信号。 在接收到编码信号时，执行逆对数压缩和时间频率或时标重建以提供音频信号。 版权所有（C）2013，JPO＆INPIT

    Voice recognition system using implicit speaker adaption
    9.
    发明专利
    Voice recognition system using implicit speaker adaption 有权
    语音识别系统使用隐含的扬声器自适应

    公开(公告)号：JP2008203876A

    公开(公告)日：2008-09-04

    申请号：JP2008101180

    申请日：2008-04-09

    Applicant: Qualcomm Inc ,  クゥアルコム・インコーポレイテッドQualcomm Incorporated

    Inventor： MALAYATH NARENDRANATH ,  DEJACO ANDREW P ,  CHANG CHIENCHUNG ,  JALIL SUHAIL ,  BI NING ,  GARUDADRI HARINATH

    IPC: G10L15/06 ,  G10L15/10 ,  G10L15/12 ,  G10L15/14 ,  G10L15/16 ,  G10L17/04

    CPC classification number: G10L15/142 ,  G10L15/07 ,  G10L15/10 ,  G10L15/12 ,  G10L15/144 ,  G10L15/32

    Abstract: PROBLEM TO BE SOLVED: To provide a voice recognition method which excellently functions before and during generation of a speaker dependent (SD) acoustic model through unsupervised training.
    SOLUTION: At least one speaker-independent (SI) acoustic model 230 or 232 is used in combination with at least one speaker-dependent (SD) acoustic model 234 to provide a level of voice recognition performance that at least equals that of the purely SI acoustic model 230 or 232. The disclosed system continually uses unsupervised training to update the acoustic templates in the one or more SD acoustic models. The system then uses the updated SD acoustic models 234 in combination with the at least one SI acoustic model 230 or 232 to provide improved voice recognition performance even during voice recognition testing.
    COPYRIGHT: (C)2008,JPO&INPIT

    Abstract translation: 要解决的问题：提供一种通过无监督训练在演讲者依赖（SD）声学模型产生之前和期间优异地起作用的语音识别方法。 解决方案：至少一个与扬声器无关的（SI）声学模型230或232与至少一个说话者相关（SD）声学模型234组合使用，以提供至少等于 纯SI声学模型230或232.所公开的系统连续地使用无监督训练来更新一个或多个SD声学模型中的声学模板。 该系统然后使用更新的SD声学模型234与至少一个SI声学模型230或232组合以提供甚至在语音识别测试期间改进的语音识别性能。 版权所有（C）2008，JPO＆INPIT

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