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31. 发明申请

US20060271357A1 Sub-band voice codec with multi-stage codebooks and redundant coding 有权
标题翻译：具有多级码本和冗余编码的子带语音编解码器
公开(公告)号：US20060271357A1
公开(公告)日：2006-11-30
申请号：US11197914
申请日：2005-08-04
申请人： Tian Wang , Kazuhito Koishida , Hosam Khalil , Xiaoqin Sun , Wei-Ge Chen
发明人： Tian Wang , Kazuhito Koishida , Hosam Khalil , Xiaoqin Sun , Wei-Ge Chen
IPC分类号： G10L19/12
CPC分类号： G10L19/005 , G10L19/12 , G10L2019/0005
摘要： 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.
摘要翻译：描述与音频信息的编码和解码相关的技术和工具。例如，用于解码当前帧的冗余编码信息包括仅与先前帧的一部分相关联的信号历史信息。作为另一示例，用于对已编码单元进行解码的冗余编码信息包括仅当前一编码单元不可用时才将用于解码当前编码单元的码本级的参数。作为另一示例，编码音频单元各自包括指示编码单元是否包括表示音频信号的段的主编码信息的字段，以及编码单元是否包括用于解码主编码信息的冗余编码信息。

32. 发明申请

US20050278172A1 Gain constrained noise suppression 有权
标题翻译：增加约束噪声抑制
公开(公告)号：US20050278172A1
公开(公告)日：2005-12-15
申请号：US10869467
申请日：2004-06-15
申请人： Kazuhito Koishida , Feng Zhuge , Hosam Khalil , Tian Wang , Wei-ge Chen
发明人： Kazuhito Koishida , Feng Zhuge , Hosam Khalil , Tian Wang , Wei-ge Chen
IPC分类号： G10L15/20 , G10L21/02
CPC分类号： G10L21/0208 , G10L21/0232
摘要： 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). First, a noisy factor is computed based on a ratio of the number of noise bins to the total number of bins for the current frame, where a zero-valued noisy factor means only using constant gain for all the spectrum values and noisy factor of one means no smoothing at all. Then, this noisy factor is used to alter the gain factors, such as by cutting off the high frequency components of the gain factors in the frequency domain.
摘要翻译：用于语音的增益约束噪声抑制更精确地估计包括在语音期间的噪声，以减少从噪声抑制引入的音乐噪声伪像。通过对语音信号的每个短时间频谱值S（m，k）应用频谱增益G（m，k）来进行噪声抑制，其中m是帧号，k是频谱索引。频谱值被分组成频率仓，并且对于被分类为“噪声仓”的每个仓估计的噪声特性。能量参数在时域和频域均被平滑，以改善每个bin的噪声估计。基于当前信号频谱和噪声估计来计算增益因子G（m，k），然后在施加到信号频谱值S（m，k）之前进行平滑处理。首先，基于噪声箱数与当前帧的总数的比率来计算噪声因子，其中零值噪声因子意味着仅对所有频谱值使用恒定增益并且噪声因子为1 意味着没有平滑。然后，这种噪声因子用于改变增益因子，例如通过切断频域中增益因子的高频分量。

33. 发明申请

US20050228651A1 Robust real-time speech codec 有权
标题翻译：强大的实时语音编解码器
公开(公告)号：US20050228651A1
公开(公告)日：2005-10-13
申请号：US10816466
申请日：2004-03-31
申请人： Tian Wang , Hosam Khalil , Kazuhito Koishida , Wei-Ge Chen , Mu Han
发明人： Tian Wang , Hosam Khalil , Kazuhito Koishida , Wei-Ge Chen , Mu Han
IPC分类号： G10L11/06 , G10L19/08
CPC分类号： G10L19/08 , G10L19/005 , G10L19/22
摘要： 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.
摘要翻译：描述了音频编解码器中的速率/质量控制和丢失弹性的各种策略。各种策略可以组合使用或独立使用。例如，实时语音编解码器使用帧内编码/解码，自适应多模式前向纠错[“FEC”]和速率/质量控制技术。帧内帧帮助解码器从分组丢失中快速恢复，而预测帧仍然强调压缩效率。描述了用于插入帧内和信令帧内/预测帧的各种策略。利用自适应多模式FEC，编码器在多种模式之间自适应地选择以有效且快速地提供考虑到当前可用于FEC的带宽的FEC级别。 FEC信息本身可以相对于主编码信息进行预测编码和解码。各种速率/质量和FEC控制策略允许对可用带宽和网络条件进行额外的调整。

34. 发明授权

US08996557B2 Query and matching for content recognition 有权
标题翻译：查询和匹配内容识别
公开(公告)号：US08996557B2
公开(公告)日：2015-03-31
申请号：US13110185
申请日：2011-05-18
申请人： Kazuhito Koishida , David Nister , Ian Simon , Tom Butcher
发明人： Kazuhito Koishida , David Nister , Ian Simon , Tom Butcher
IPC分类号： G06F17/30
CPC分类号： G06F17/30743
摘要： Various embodiments enable audio data, such as music data, to be captured, by a device, from a background environment and processed to formulate a query that can then be transmitted to a content recognition service. In one or more embodiments, multiple queries are transmitted to the content recognition service. In at least some embodiments, subsequent queries can progressively incorporate previous queries plus additional data that is captured. In one or more embodiments, responsive to receiving the query, the content recognition service can employ a multi-stage matching technique to identify content items responding to the query. This matching technique can be employed as queries are progressively received.
摘要翻译：各种实施例使得诸如音乐数据的音频数据能够被设备从背景环境中捕获并被处理以制定可以被发送到内容识别服务的查询。在一个或多个实施例中，多个查询被发送到内容识别服务。在至少一些实施例中，后续查询可以逐渐地并入先前查询加上所捕获的附加数据。在一个或多个实施例中，响应于接收查询，内容识别服务可以采用多阶段匹配技术来识别响应于查询的内容项目。可以采用这种匹配技术，因为逐渐接收到查询。

35. 发明授权

US08645146B2 Bitstream syntax for multi-process audio decoding 有权
标题翻译：多进程音频解码的比特流语法
公开(公告)号：US08645146B2
公开(公告)日：2014-02-04
申请号：US13595939
申请日：2012-08-27
申请人： Kazuhito Koishida , Sanjeev Mehrotra , Chao He , Wei-Ge Chen
发明人： Kazuhito Koishida , Sanjeev Mehrotra , Chao He , Wei-Ge Chen
IPC分类号： G10L19/00
CPC分类号： G10L19/167 , G10L19/002 , G10L19/008 , G10L19/022 , G10L19/03 , G10L19/038 , G10L19/04 , G10L19/24
摘要： An audio decoder provides a combination of decoding components including components implementing base band decoding, spectral peak decoding, frequency extension decoding and channel extension decoding techniques. The audio decoder decodes a compressed bitstream structured by a bitstream syntax scheme to permit the various decoding components to extract the appropriate parameters for their respective decoding technique.
摘要翻译：音频解码器提供包括实现基带解码，频谱峰值解码，频率扩展解码和信道扩展解码技术的组件的解码组件的组合。音频解码器解码由比特流语法方案构成的压缩比特流，以允许各种解码组件为它们各自的解码技术提取适当的参数。

36. 发明授权

US07774205B2 Coding of sparse digital media spectral data 有权
标题翻译：稀疏数字媒体光谱数据编码
公开(公告)号：US07774205B2
公开(公告)日：2010-08-10
申请号：US11764108
申请日：2007-06-15
申请人： Kazuhito Koishida , Sanjeev Mehrotra , Wei-Ge Chen
发明人： Kazuhito Koishida , Sanjeev Mehrotra , Wei-Ge Chen
IPC分类号： G10L21/04
CPC分类号： G10L19/02 , G10L19/0212 , G10L19/032 , G10L19/18
摘要： An audio encoder/decoder provides efficient compression of spectral transform coefficient data characterized by sparse spectral peaks. The audio encoder/decoder applies a temporal prediction of the frequency position of spectral peaks. The spectral peaks in the transform coefficients that are predicted from those in a preceding transform coding block are encoded as a shift in frequency position from the previous transform coding block and two non-zero coefficient levels. The prediction may avoid coding very large zero-level transform coefficient runs as compared to conventional run length coding. For spectral peaks not predicted from those in a preceding transform coding block, the spectral peaks are encoded as a value trio of a length of a run of zero-level spectral transform coefficients, and two non-zero coefficient levels.
摘要翻译：音频编码器/解码器提供以稀疏频谱峰值为特征的频谱变换系数数据的有效压缩。音频编码器/解码器对频谱峰值的频率位置进行时间预测。从前一变换编码块中预测的变换系数中的频谱峰值被编码为来自先前变换编码块和两个非零系数电平的频率位置的移位。与常规游程长度编码相比，预测可以避免编码非常大的零电平变换系数运行。对于未在前面的变换编码块中预测的频谱峰值，频谱峰值被编码为零电平频谱变换系数的行程的长度和两个非零系数电平的三值。

37. 发明申请

US20100125455A1 AUDIO ENCODING AND DECODING WITH INTRA FRAMES AND ADAPTIVE FORWARD ERROR CORRECTION 审中-公开
标题翻译：音频编码和解码与内部框架和自适应前向错误校正
公开(公告)号：US20100125455A1
公开(公告)日：2010-05-20
申请号：US12692417
申请日：2010-01-22
申请人： Tian Wang , Hosam A. Khalil , Kazuhito Koishida , Wei-Ge Chen , Mu Han
发明人： Tian Wang , Hosam A. Khalil , Kazuhito Koishida , Wei-Ge Chen , Mu Han
IPC分类号： G10L19/08
CPC分类号： G10L19/08 , G10L19/005 , G10L19/22
摘要： 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.
摘要翻译：描述了音频编解码器中的速率/质量控制和丢失弹性的各种策略。各种策略可以组合使用或独立使用。例如，实时语音编解码器使用帧内编码/解码，自适应多模式前向纠错[“FEC”]和速率/质量控制技术。帧内帧帮助解码器从分组丢失中快速恢复，而预测帧仍然强调压缩效率。描述了用于插入帧内和信令帧内/预测帧的各种策略。利用自适应多模式FEC，编码器在多种模式之间自适应地选择以有效且快速地提供考虑到当前可用于FEC的带宽的FEC级别。 FEC信息本身可以相对于主编码信息进行预测编码和解码。各种速率/质量和FEC控制策略允许对可用带宽和网络条件进行额外的调整。

38. 发明授权

US07707034B2 Audio codec post-filter 有权
标题翻译：音频编解码后置滤波器
公开(公告)号：US07707034B2
公开(公告)日：2010-04-27
申请号：US11142603
申请日：2005-05-31
申请人： Xiaoqin Sun , Tian Wang , Hosam A. Khalil , Kazuhito Koishida , Wei-Ge Chen
发明人： Xiaoqin Sun , Tian Wang , Hosam A. Khalil , Kazuhito Koishida , Wei-Ge Chen
IPC分类号： G10L19/00
CPC分类号： G10L19/26
摘要： 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.
摘要翻译：描述了处理重建音频信号的技术和工具。例如，使用至少部分地在频域中计算的滤波器系数在时域中对重构音频信号进行滤波。作为另一示例，产生用于对重构音频信号进行滤波的滤波器系数的集合包括限定一组系数值的一个或多个峰值。作为另一示例，对于子带编解码器，在两个子带之间的交叉点附近的频率区域中，重构的复合信号被增强。

39. 发明授权

US07668712B2 Audio encoding and decoding with intra frames and adaptive forward error correction 有权
标题翻译：音频编码和解码与帧内和自适应前向纠错
公开(公告)号：US07668712B2
公开(公告)日：2010-02-23
申请号：US10816466
申请日：2004-03-31
申请人： Tian Wang , Hosam A. Khalil , Kazuhito Koishida , Wei-Ge Chen , Mu Han
发明人： Tian Wang , Hosam A. Khalil , Kazuhito Koishida , Wei-Ge Chen , Mu Han
IPC分类号： G10L19/00
CPC分类号： G10L19/08 , G10L19/005 , G10L19/22
摘要： 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.
摘要翻译：描述了音频编解码器中的速率/质量控制和丢失弹性的各种策略。各种策略可以组合使用或独立使用。例如，实时语音编解码器使用帧内编码/解码，自适应多模式前向纠错[“FEC”]和速率/质量控制技术。帧内帧帮助解码器从分组丢失中快速恢复，而预测帧仍然强调压缩效率。描述了用于插入帧内和信令帧内/预测帧的各种策略。利用自适应多模式FEC，编码器在多种模式之间自适应地选择以有效且快速地提供考虑到当前可用于FEC的带宽的FEC级别。 FEC信息本身可以相对于主编码信息进行预测编码和解码。各种速率/质量和FEC控制策略允许对可用带宽和网络条件进行额外的调整。

40. 发明申请

US20080040121A1 Sub-band voice codec with multi-stage codebooks and redundant coding 有权
标题翻译：具有多级码本和冗余编码的子带语音编解码器
公开(公告)号：US20080040121A1
公开(公告)日：2008-02-14
申请号：US11973690
申请日：2007-10-09
申请人： Tian Wang , Kazuhito Koishida , Hosam Khalil , Xiaoqin Sun , Wei-Ge Chen
发明人： Tian Wang , Kazuhito Koishida , Hosam Khalil , Xiaoqin Sun , Wei-Ge Chen
IPC分类号： G10L19/00
CPC分类号： G10L19/005 , G10L19/12 , G10L2019/0005
摘要： 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.
摘要翻译：描述与音频信息的编码和解码相关的技术和工具。例如，用于解码当前帧的冗余编码信息包括仅与先前帧的一部分相关联的信号历史信息。作为另一示例，用于对已编码单元进行解码的冗余编码信息包括仅当前一编码单元不可用时才将用于解码当前编码单元的码本级的参数。作为另一示例，编码音频单元各自包括指示编码单元是否包括表示音频信号的段的主编码信息的字段，以及编码单元是否包括用于解码主编码信息的冗余编码信息。

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