专利快速检索-快速检索全球专利，免费商用专利数据库-IPRDB

1. 发明授权

US06636829B1 Speech communication system and method for handling lost frames 有权
标题翻译：用于处理丢帧的语音通信系统和方法
公开(公告)号：US06636829B1
公开(公告)日：2003-10-21
申请号：US09617191
申请日：2000-07-14
申请人： Adil Benyassine , Eyal Shlomot , Huan-Yu Su
发明人： Adil Benyassine , Eyal Shlomot , Huan-Yu Su
IPC分类号： G10L1900
CPC分类号： G10L19/08 , G10L19/005 , G10L19/07 , G10L19/083 , G10L25/90 , G10L2019/0012
摘要： An exemplary decoder comprises a receiver that receives parameters of a speech signal on a frame-by-frame basis, a control logic for decoding parameters and for resynthesizing the speech signal, the control logic including a minimum spacing indicative of a minimum difference required between LSFs of consecutive frames, a frame recovery logic that, when a lost frame detector detects a lost frame, sets the minimum spacing for the lost frame to a first value which is greater than the minimum spacing for the previously received frame, and/or uses pitch lag parameters of a plurality of previously received frames to extrapolate a pitch lag parameter for the lost frame, and/or sets gain parameter of a subframe of the lost frame in a first manner if the lost gain parameter is an adaptive codebook gain parameter and in a second manner if the lost gain parameter is a fixed codebook gain parameter.
摘要翻译：示例性解码器包括接收器，其逐帧地接收语音信号的参数，用于解码参数并用于再合成语音信号的控制逻辑，所述控制逻辑包括指示LSF之间所需的最小差异的最小间隔连续帧的帧恢复逻辑，当丢失帧检测器检测到丢失帧时，将丢失帧的最小间隔设置为大于先前接收帧的最小间隔的第一值，和/或使用间距多个先前接收的帧的滞后参数，以推断丢失帧的音调滞后参数，和/或以丢失的增益参数为自适应码本增益参数，以第一种方式设置丢失帧的子帧的增益参数，并且丢失增益参数是固定码本增益参数的第二种方式。

2. 发明授权

US5920853A Signal compression using index mapping technique for the sharing of quantization tables 失效
标题翻译：信号压缩使用索引映射技术共享量化表
公开(公告)号：US5920853A
公开(公告)日：1999-07-06
申请号：US702780
申请日：1996-08-23
申请人： Adil Benyassine , Huan-Yu Su , Eyal Shlomot
发明人： Adil Benyassine , Huan-Yu Su , Eyal Shlomot
IPC分类号： H04N7/26 , G06T9/00 , H03M7/30 , G06F17/30 , G06F5/00
CPC分类号： G06T9/008 , H03M7/3082 , Y10S707/99931
摘要： A signal compression system includes a coder and a decoder. The coder includes an extract unit for extracting an input feature vector from an input signal, a coder memory unit for storing a predesigned vector quantization (VQ) table for the coder such that the coder memory unit uses a set of primary indices to address entries within the pre-designed VQ table, a coder mapping unit for mapping indices from a set of secondary indices to the first set of indices, and a search unit for searching for one index out of the set of secondary indices, wherein the index from the set of secondary indices corresponds to an entry in the coder memory unit, and the entry best represents the input feature vector according to some predetermined criteria. On the decoder side, the decoder includes a decoder memory unit for storing the same pre-designed VQ table and set of primary indices as the coder memory unit, a decoder mapping unit, and a retrieval unit, wherein the entry indicated by the index best represents the input feature vector.
摘要翻译：信号压缩系统包括编码器和解码器。编码器包括用于从输入信号提取输入特征向量的提取单元，编码器存储单元，用于存储用于编码器的预先设计的矢量量化（VQ）表，使得编码器存储单元使用一组主要索引来寻址预先设计的VQ表，用于映射从一组二次索引到第一组索引的索引的编码器映射单元，以及用于搜索该次要索引集合中的一个索引的搜索单元，其中来自该集合的索引次要索引对应于编码器存储单元中的条目，并且条目最好地表示根据某些预定标准的输入特征向量。在解码器侧，解码器包括解码器存储器单元，用于存储与编码器存储单元相同的预先设计的VQ表和一组主要索引，解码器映射单元和检索单元，其中由索引最佳指示的条目代表输入特征向量。

3. 发明授权

US07080010B2 Complexity resource manager for multi-channel speech processing 有权
标题翻译：用于多声道语音处理的复杂性资源管理器
公开(公告)号：US07080010B2
公开(公告)日：2006-07-18
申请号：US10911118
申请日：2004-08-03
申请人： Eyal Shlomot , Huan-Yu Su
发明人： Eyal Shlomot , Huan-Yu Su
IPC分类号： G10L19/02
CPC分类号： G10L15/285
摘要： A multi-channel speech processor for encoding speech in a packet network environment is disclosed. In one illustrative aspect, a complexity resource manager (CRM) is executed by a controller or processor. The CRM manages the level of complexity of encoding which is used by a signal processing unit (SPU) to convert the speech signal into packet data. In general, the CRM determines the level of complexity of encoding based on a calculated complexity budget, where the complexity budget is determined based on the time required to process prior speech signal channels and the time available to process the remaining channels. In this way, the CRM is able to control the overall complexity of the speech processor through its ability to signal the SPU to encode speech signal in a complexity reduced mode based on the calculated complexity budget under certain conditions.
摘要翻译：公开了一种用于在分组网络环境中编码语音的多声道语音处理器。在一个说明性方面，复杂性资源管理器（CRM）由控制器或处理器执行。 CRM管理由信号处理单元（SPU）用于将语音信号转换成分组数据的编码的复杂程度。通常，CRM基于计算的复杂度预算确定编码的复杂程度，其中基于处理先前语音信号信道所需的时间和可用于处理剩余信道的时间来确定复杂度预算。以这种方式，CRM能够通过其在特定条件下基于计算的复杂度预算在复杂度降低模式下对SPU进行信号编码语音信号的能力来控制语音处理器的总体复杂性。

4. 发明授权

US07146309B1 Deriving seed values to generate excitation values in a speech coder 有权
标题翻译：导出种子值以在语音编码器中产生激励值
公开(公告)号：US07146309B1
公开(公告)日：2006-12-05
申请号：US10653874
申请日：2003-09-02
申请人： Adil Benyassine , Eyal Shlomot , Huan-Yu Su
发明人： Adil Benyassine , Eyal Shlomot , Huan-Yu Su
IPC分类号： G10L19/00
CPC分类号： G10L19/08
摘要： There are provided methods and devices for generating excitation values for a speech signal. In one aspect, an example method comprises obtaining one or more characteristics of a first speech frame of the speech signal, deriving a first seed value based on the one or more characteristics of the first speech frame, providing the first seed value to a Gaussian time series generator; and using the Gaussian time series generator to generate an excitation values for the first frame. The one or more characteristics may include a spectrum information of the first frame, an energy information of the first frame, or a gain information of the first frame.
摘要翻译：提供了用于产生语音信号的激励值的方法和装置。在一个方面，示例性方法包括获得语音信号的第一语音帧的一个或多个特征，基于第一语音帧的一个或多个特征导出第一种子值，将第一种子值提供给高斯时间串联发电机; 并使用高斯时间序列发生器来产生第一帧的激励值。一个或多个特征可以包括第一帧的频谱信息，第一帧的能量信息或第一帧的增益信息。

5. 发明授权

US06789058B2 Complexity resource manager for multi-channel speech processing 有权
标题翻译：用于多声道语音处理的复杂性资源管理器
公开(公告)号：US06789058B2
公开(公告)日：2004-09-07
申请号：US10271576
申请日：2002-10-15
申请人： Eyal Shlomot , Huan-Yu Su
发明人： Eyal Shlomot , Huan-Yu Su
IPC分类号： G10L1900
CPC分类号： G10L15/285
摘要： A multi-channel speech processor for encoding speech in a packet network environment is disclosed. In one illustrative aspect, a complexity resource manager (CRM) is executed by a controller or processor. The CRM manages the level of complexity of encoding which is used by a signal processing unit (SPU) to convert the speech signal into packet data. In general, the CRM determines the level of complexity of encoding based on a calculated complexity budget, where the complexity budget is determined based on the time required to process prior speech signal channels and the time available to process the remaining channels. In this way, the CRM is able to control the overall complexity of the speech processor through its ability to signal the SPU to encode speech signal in a complexity reduced mode based on the calculated complexity budget under certain conditions.
摘要翻译：公开了一种用于在分组网络环境中编码语音的多声道语音处理器。在一个说明性方面，复杂性资源管理器（CRM）由控制器或处理器执行。 CRM管理由信号处理单元（SPU）用于将语音信号转换成分组数据的编码的复杂程度。通常，CRM基于计算的复杂度预算确定编码的复杂程度，其中基于处理先前语音信号信道所需的时间和可用于处理剩余信道的时间来确定复杂度预算。以这种方式，CRM能够通过其在特定条件下基于计算的复杂度预算在复杂度降低模式下对SPU进行信号编码语音信号的能力来控制语音处理器的总体复杂性。

6. 发明授权

US06463414B1 Conference bridge processing of speech in a packet network environment 有权
标题翻译：会议桥处理语音在分组网环境中
公开(公告)号：US06463414B1
公开(公告)日：2002-10-08
申请号：US09547832
申请日：2000-04-12
申请人： Huan-Yu Su , Eyal Shlomot , Jes Thyssen , Adil Benyassine , Yang Gao
发明人： Huan-Yu Su , Eyal Shlomot , Jes Thyssen , Adil Benyassine , Yang Gao
IPC分类号： G10L1102
CPC分类号： G10L19/173
摘要： There is provided a conference bridge or transcoder configured to intelligently handle multiple speech channels in the contest of a packet network, wherein various speech channels may adhere to variety of speech encoding standards. For example, the conference bridge establishes framing and alignment of multiple incoming speech channels associated with multiple participants, extracts parameters from the speech samples, mixes the parameters, and re-encodes the resulting speech samples for transmission to the participants. In one aspect, a speech processing method comprises decoding a first bitstream according to a first coding scheme to generate first speech samples and a first side information; generating second speech samples and a second side information using the first speech samples and the first side information, for use according to a second coding scheme; and creating a second bitstream, encoded based on the second coding scheme, using the second speech samples and the second side information.
摘要翻译：提供了一种配置成在分组网络的比赛中智能地处理多个语音信道的会议桥或代码转换器，其中各种语音信道可以遵循各种语音编码标准。例如，会议桥建立与多个参与者相关联的多个输入语音信道的成帧和对准，从语音样本中提取参数，混合参数，并对所得到的语音样本进行重新编码以传输给参与者。一方面，语音处理方法包括根据第一编码方案对第一比特流进行解码，以产生第一语音样本和第一侧信息; 使用第一语音样本和第一侧信息生成第二语音样本和第二侧信息，以便根据第二编码方案使用; 以及使用所述第二语音样本和所述第二侧信息来创建基于所述第二编码方案编码的第二比特流。

7. 发明申请

US20050010405A1 Complexity resource manager for multi-channel speech processing 有权
标题翻译：用于多声道语音处理的复杂性资源管理器
公开(公告)号：US20050010405A1
公开(公告)日：2005-01-13
申请号：US10911118
申请日：2004-08-03
申请人： Eyal Shlomot , Huan-Yu Su
发明人： Eyal Shlomot , Huan-Yu Su
IPC分类号： G10L15/28 , G10L19/02
CPC分类号： G10L15/285
摘要： A multi-channel speech processor for encoding speech in a packet network environment is disclosed. In one illustrative aspect, a complexity resource manager (CRM) is executed by a controller or processor. The CRM manages the level of complexity of encoding which is used by a signal processing unit (SPU) to convert the speech signal into packet data. In general, the CRM determines the level of complexity of encoding based on a calculated complexity budget, where the complexity budget is determined based on the time required to process prior speech signal channels and the time available to process the remaining channels. In this way, the CRM is able to control the overall complexity of the speech processor through its ability to signal the SPU to encode speech signal in a complexity reduced mode based on the calculated complexity budget under certain conditions.
摘要翻译：公开了一种用于在分组网络环境中编码语音的多声道语音处理器。在一个说明性方面，复杂性资源管理器（CRM）由控制器或处理器执行。 CRM管理由信号处理单元（SPU）用于将语音信号转换成分组数据的编码的复杂程度。通常，CRM基于计算的复杂度预算确定编码的复杂程度，其中基于处理先前语音信号信道所需的时间和可用于处理剩余信道的时间来确定复杂度预算。以这种方式，CRM能够通过其在特定条件下基于计算的复杂度预算在复杂度降低模式下对SPU进行信号编码语音信号的能力来控制语音处理器的总体复杂性。

8. 发明授权

US06721712B1 Conversion scheme for use between DTX and non-DTX speech coding systems 有权
标题翻译： DTX与非DTX语音编码系统之间的转换方案
公开(公告)号：US06721712B1
公开(公告)日：2004-04-13
申请号：US10057250
申请日：2002-01-24
申请人： Adil Benyassine , Eyal Shlomot , Huan-Yu Su
发明人： Adil Benyassine , Eyal Shlomot , Huan-Yu Su
IPC分类号： G10L1900
CPC分类号： H04W88/181 , G10L19/173
摘要： In an exemplary conversion scheme, a frame of a first speech signal comprising a plurality of frames encoded at a plurality of first rates, including a first non-speech rate, is received. The rate of the received frame is determined, and if the received frame is encoded at the first non-speech rate, then the received frame is re-encoded at either a second or third non-speech rate to generate a frame of a second speech signal. Moreover, a system for converting a speech signal comprises a receiver for receiving a frame of a first speech signal and a processor capable of determining the encoding rate of the received frame and re-encoding the received frame at either a second or third non-speech rate if the received frame was originally encoded at a first non-speech rate.
摘要翻译：在示例性转换方案中，接收包括以包括第一非语音速率在内的多个第一速率编码的多个帧的第一语音信号的帧。确定接收帧的速率，并且如果以第一非语音速率对接收的帧进行编码，则接收的帧以第二或第三非语音速率被重新编码，以产生第二语音的帧信号。此外，用于转换语音信号的系统包括用于接收第一语音信号的帧的接收机和能够确定接收到的帧的编码速率并且在第二或第三非语音中重新编码接收的帧的处理器如果接收的帧最初以第一非语音速率被编码，则速率。

9. 发明授权

US10366701B1 Adaptive multi-microphone beamforming 有权
公开(公告)号：US10366701B1
公开(公告)日：2019-07-30
申请号：US15681395
申请日：2017-08-20
申请人： Huan-Yu Su
发明人： Huan-Yu Su
IPC分类号： G10L15/20 , G10L21/0216 , G10L21/0224 , H04R1/32 , H04R29/00 , G10L21/0264
摘要： Provided is a method and computer program product for producing an enhanced audio signal for an output device from audio signals received by 2 or more microphones in close proximity to each other. For example, one embodiment of the present invention comprises the steps of receiving a first input audio signal from the first microphone, digitizing the first input audio signal to produce a first digitized audio input signal, receiving a second input audio input signal from the second microphone, digitizing the second input audio input signal to produce a second digitized audio input signal, using the first digitized audio input signal as a reference signal to an adaptive prediction filter, using the second digitized audio input signal as input to said adaptive prediction filter and finally adding a prediction result signal from the adaptive prediction filter to the first digitized audio input signal to produce the enhanced audio signal. In other embodiments, any number of microphones can be used, and in all embodiments there is no requirement to detect or locate the source or direction of arrival of the input audio signals.

10. 发明授权

US07796623B2 Detecting and reporting a loss of connection by a telephone 有权
标题翻译：通过电话检测和报告连接丢失
公开(公告)号：US07796623B2
公开(公告)日：2010-09-14
申请号：US12384019
申请日：2009-03-30
申请人： Michael M. Metzger , Huan-Yu Su , Armin Abold
发明人： Michael M. Metzger , Huan-Yu Su , Armin Abold
IPC分类号： H04L12/28 , H04M3/22
CPC分类号： H04M3/2236 , H04L41/5009 , H04L41/5087 , H04L65/80 , H04M7/1205
摘要： There is provided a method of detecting and reporting poor voice quality for use by a gateway device. The method comprises facilitating a connection between a telephone and a remote telephone via a network, and detecting a poor voice quality indictor during the connection. The method further comprises capturing, for a pre-determined period of time, telephone voice data being exchanged between the gateway and the telephone, network voice data being exchanged between the gateway and the network, and gateway parameters. The method also comprises packetizing the telephone voice data, the network voice data and the gateway parameters into a plurality packets having a network address of a network storage, and transmitting the plurality packets destined for the network storage via the network. In one aspect, the poor voice quality indictor may be generated by a user of the telephone in response to a poor voice quality of the connection.
摘要翻译：提供了一种检测和报告由网关设备使用的较差语音质量的方法。该方法包括通过网络促进电话和远程电话之间的连接，以及在连接期间检测不良语音质量指示符。该方法还包括：在预定时间段内，捕获在网关与电话之间交换的电话语音数据，网关和网络之间交换的网络语音数据以及网关参数。该方法还包括将电话语音数据，网络语音数据和网关参数分组成具有网络存储器的网络地址的多个分组，并且经由网络发送去往网络存储的多个分组。在一个方面，响应于连接的差的语音质量，可能由电话的用户产生差的语音质量指示符。

你已经成功收藏专利！

检索式保存成功!

IPRDB

热门服务

关于我们

友情链接

联系方式