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1. 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。

2. 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。

3. 发明申请

US20060271354A1 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：US20060271354A1
公开(公告)日：2006-11-30
申请号：US11142603
申请日：2005-05-31
申请人： Xiaoqin Sun , Tian Wang , Hosam Khalil , Kazuhito Koishida , Wei-Ge Chen
发明人： Xiaoqin Sun , Tian Wang , Hosam Khalil , Kazuhito Koishida , Wei-Ge Chen
IPC分类号： G10L19/14
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。

4. 发明申请

US20060271373A1 Robust decoder 有权
公开(公告)号：US20060271373A1
公开(公告)日：2006-11-30
申请号：US11142602
申请日：2005-05-31
申请人： Hosam Khalil , Tian Wang , Kazuhito Koishida , Xiaoqin Sun , Wei-Ge Chen
发明人： Hosam Khalil , Tian Wang , Kazuhito Koishida , Xiaoqin Sun , Wei-Ge Chen
IPC分类号： G10L21/00
CPC分类号： G10L21/045 , G10L19/005
摘要： 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.

5. 发明申请

US20060271359A1 Robust decoder 有权
公开(公告)号：US20060271359A1
公开(公告)日：2006-11-30
申请号：US11197792
申请日：2005-08-04
申请人： Hosam Khalil , Tian Wang , Kazuhito Koishida , Xiaoqin Sun , Wei-Ge Chen
发明人： Hosam Khalil , Tian Wang , Kazuhito Koishida , Xiaoqin Sun , Wei-Ge Chen
IPC分类号： G10L21/02
CPC分类号： G10L21/045 , G10L19/005
摘要： 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.

6. 发明申请

US20060271355A1 SUB-BAND VOICE CODEC WITH MULTI-STAGE CODEBOOKS AND REDUNDANT CODING 有权
标题翻译：具有多级代码和冗余编码的子带语音编解码器
公开(公告)号：US20060271355A1
公开(公告)日：2006-11-30
申请号：US11142605
申请日：2005-05-31
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。

7. 发明申请

US20080040105A1 Sub-band voice codec with multi-stage codebooks and redundant coding 有权
公开(公告)号：US20080040105A1
公开(公告)日：2008-02-14
申请号：US11973689
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： G10L19/12
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.

8. 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。

9. 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。

10. 发明申请

US20130006622A1 ADAPTIVE CONFERENCE COMFORT NOISE 有权
标题翻译：适应会议舒适的噪音
公开(公告)号：US20130006622A1
公开(公告)日：2013-01-03
申请号：US13170567
申请日：2011-06-28
申请人： Hosam Khalil , Xiaoqin Sun , Hong Wang Sodoma , Warren Lam
发明人： Hosam Khalil , Xiaoqin Sun , Hong Wang Sodoma , Warren Lam
IPC分类号： G10L15/20
CPC分类号： G10L19/012 , H04M3/568 , H04M2203/352
摘要： A continuous comfort noise is provided that is overlaid for the entire duration of a conference call scenario. The comfort noise may be adapted to match the levels of the actual background noise detected on one or more of the conference call participant's devices on the transmitting end(s) of a conference call as well as the participants' speech levels. The comfort noise may also be adapted to the type of listening device employed on the receiving end of a conference call. The comfort noise level may be customized to an appropriate and comfortable level for the type of listening device being used, and the system may continuously mix the comfort noise with incoming audio signals for the entire duration of a conference call, lowering the comfort noise level gradually during speaking periods for additional user experience improvement.
摘要翻译：提供连续的舒适噪声，其在电话会议场景的整个持续时间内被覆盖。舒适噪声可以适于匹配在电话会议的发送端上的一个或多个电话会议参与者设备上检测到的实际背景噪声的电平以及参与者的语音电平。舒适噪声也可以适应于在电话会议的接收端采用的听音装置的类型。可以将舒适噪声水平定制为适用于所使用的听音装置类型的舒适水平，并且系统可以在电话会议的整个持续时间内将舒适噪声与输入音频信号连续混合，逐渐降低舒适噪声电平在演讲期间，为了增加用户体验。

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