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1. 发明授权

US07206404B2 Communications system and method utilizing centralized signal processing 有权
标题翻译：采用集中式信号处理的通信系统和方法
公开(公告)号：US07206404B2
公开(公告)日：2007-04-17
申请号：US11124772
申请日：2005-05-09
申请人： James H. Parry , Peter Hsiang
发明人： James H. Parry , Peter Hsiang
IPC分类号： H04M9/08
CPC分类号： H04M9/08
摘要： A communications system and method performs centralized signal processing on received audio signals. A plurality of terminals are coupled to a processing switch via links. The terminals can be, for example, dedicated speakerphones, desktop handsets, or personal computers with audio capabilities. The links can be wired or wireless, can carry analog or digital signals, and can be shared with other users in a network. The switch receives the audio data from the terminals, processes the data according to desired acoustical procedures, creates one or more output mixes, and provides the output mixes to the appropriate terminals. The operation of the processing switch is controlled by a communications support module (CSM) which can receive, process, and send data to/from multiple terminals simultaneously. The CSM receives audio signals from the terminals. The CSM uses stored room models holding room model information including data and/or filters representing the acoustic properties of the terminal and/or the environment surrounding the terminal to produce the audio signals. Signal processing (SP) modules provide a pool of SP power from which the CSM can draw to process audio signals received from or being sent to the terminals. The CSM uses the SP modules to perform signal processing including acoustic echo cancellation, automatic gain control, noise reduction. The CSM also uses a mixing module to perform signal mixing.
摘要翻译：通信系统和方法对接收到的音频信号进行集中信号处理。多个终端通过链路耦合到处理交换机。终端可以是例如具有音频功能的专用扬声器电话，台式手持机或个人计算机。链路可以是有线或无线，可以携带模拟或数字信号，并且可以与网络中的其他用户共享。交换机从终端接收音频数据，根据期望的声学过程处理数据，创建一个或多个输出混合，并将输出混合提供给适当的终端。处理开关的操作由通信支持模块（CSM）控制，该模块可以同时从多个终端接收，处理和发送数据。 CSM从终端接收音频信号。 CSM使用保存房间模型信息的存储房间模型，包括表示终端的声学特性的数据和/或滤波器和/或终端周围的环境以产生音频信号。信号处理（SP）模块提供SP功率池，CSM可从该池中抽取来处理从终端接收或发送到终端的音频信号。 CSM使用SP模块执行信号处理，包括声学回波消除，自动增益控制，降噪。 CSM还使用混合模块来执行信号混合。

2. 发明授权

US06751404B1 Method and apparatus for detecting processor congestion during audio and video decode 失效
标题翻译：用于在音频和视频解码期间检测处理器拥塞的方法和装置
公开(公告)号：US06751404B1
公开(公告)日：2004-06-15
申请号：US09548898
申请日：2000-04-13
申请人： Paul H. Whitford , James H. Parry , See-Mong Tan
发明人： Paul H. Whitford , James H. Parry , See-Mong Tan
IPC分类号： H04N591
CPC分类号： G06F9/50 , H04N5/4401 , H04N5/602 , Y02D10/22
摘要： Disclosed is a method and system for detecting processor congestion during decompression of a stream of video and audio data. The system and method includes a processor receiving and decoding a first frame of audio data in accordance with an audio decode software algorithm. The processor generates a first audio time stamp ATS1 indicating the time at which the processor finishes decoding the first frame of audio data. Subsequently, the processor receives and decodes a second frame of audio data in accordance with the same audio decode software algorithm and generates a second audio time stamp ATS2 indicating the time at which the processor finishes decoding the second audio data frame. The first audio time stamp ATS1 is added to a predetermined amount of time T, the result of which is compared with ATS2. T, in one embodiment, is the time it takes a speaker to generate audio corresponding to a decoded frame of audio data. If ATS2 is later in time than (ATS1+T) by a predetermined amount TMIN, a signal is generated indicating that the processor is not decoding received audio frames fast enough due to processor congestion. In response to this signal, the processor workload can be redistributed in favor of increased audio decoding. If ATS2 is not later in time than (ATS1+T) by a predetermined amount TMIN, then no modifications to the processor workload need be made.
摘要翻译：公开了一种用于在解压缩视频和音频数据流期间检测处理器拥塞的方法和系统。该系统和方法包括根据音频解码软件算法接收和解码第一帧音频数据的处理器。处理器产生指示处理器完成对音频数据的第一帧的解码的时间的第一音频时间戳ATS1。随后，处理器根据相同的音频解码软件算法接收和解码第二帧音频数据，并产生指示处理器完成对第二音频数据帧的解码的时间的第二音频时间戳ATS2。将第一音频时间戳ATS1添加到预定量的时间T，将其结果与ATS2进行比较。在一个实施例中，T是扬声器产生与解码的音频数据帧相对应的音频所花费的时间。如果ATS2的时间晚于（ATS1 + T）预定量TMIN，则产生指示处理器由于处理器拥塞而足够快地解码接收的音频帧的信号。响应于该信号，可以重新分配处理器工作负载以有利于增加的音频解码。如果ATS2的时间不晚于（ATS1 + T）预定量TMIN，则不需要对处理器工作量进行任何修改。

3. 发明授权

US06650701B1 Apparatus and method for controlling an acoustic echo canceler 有权
标题翻译：用于控制声学回波消除器的装置和方法
公开(公告)号：US06650701B1
公开(公告)日：2003-11-18
申请号：US09483601
申请日：2000-01-14
申请人： Peter C. Hsiang , James H. Parry
发明人： Peter C. Hsiang , James H. Parry
IPC分类号： H03H730
CPC分类号： H03H21/0012 , H03H2021/007 , H03H2021/0074
摘要： A method and associated apparatus for controlling an acoustic canceler (“AEC”) are disclosed. Prior to passing audio signals to the AEC, a distortion detector is used to determine if the signals are distorted. If so, the AEC does not adapt is filter coefficients to the distorted signals. This technique improves the AEC's ability to adapt its filter coefficients to subsequent undistorted signals. For example, near-end or far-end audio signals above a predetermined threshold value are detected by a distortion detector which disables adaptive filter control logic so that distorted signals do not result in generation of erroneous filter coefficients.
摘要翻译：公开了一种用于控制声消除器的方法和相关装置（“AEC”）。在将音频信号传送到AEC之前，使用失真检测器来确定信号是否失真。如果是这样，AEC不适应是失真信号的滤波器系数。该技术提高了AEC将滤波系数适应于后续无失真信号的能力。例如，高于预定阈值的近端或远端音频信号由失真检测器检测，该失真检测器禁用自适应滤波器控制逻辑，使得失真信号不导致错误滤波器系数的产生。

4. 发明授权

US09325854B1 Structure and method for echo reduction without loss of information 有权
公开(公告)号：US09325854B1
公开(公告)日：2016-04-26
申请号：US13803696
申请日：2013-03-14
申请人： James H. Parry
发明人： James H. Parry
IPC分类号： H04M9/08 , H04M1/60 , G10L25/87
CPC分类号： H04M9/082 , G10L25/87 , H04M1/6016
摘要： An echo reduction method stores a received audio information stream. A sound detection flag is activated following detection of locally generated sound. Output based on the received audio information stream is muted in response to the activating the sound detection flag. Rendering status of the received audio information stream is saved, in response to the activating the sound detection flag, to reduce loss of audio information. At least a portion of the stored received audio information stream is rendered following inactivation of the sound detection flag.

5. 发明授权

US08027462B1 Structure and method for conversation like rendering for echo reduction without loss of information 有权
标题翻译：用于会话的结构和方法，如渲染回声，而不会丢失信息
公开(公告)号：US08027462B1
公开(公告)日：2011-09-27
申请号：US11502943
申请日：2006-08-11
申请人： James H. Parry
发明人： James H. Parry
IPC分类号： H04M1/00
CPC分类号： H04M1/6016 , G10L25/87
摘要： A method for conversation like rendering of a stored audio information stream determines a first location in the stored audio information stream. The first location represents a point in time when the sound detection flag became active. The method next moves from the first location to a second location in the stored audio information stream. The second location is selected based upon a criterion to make playback of the stored audio information stream appear like actual conversation. Finally, the stored audio information stream is rendered starting with audio information stored at the second location.
摘要翻译：用于会话的方法，如存储的音频信息流的呈现确定所存储的音频信息流中的第一位置。第一个位置表示声音检测标志变为活动的时间点。该方法接下来从存储的音频信息流中的第一位置移动到第二位置。基于使存储的音频信息流的播放与实际对话一样显示的标准来选择第二位置。最后，以存储在第二位置的音频信息开始存储所存储的音频信息流。

6. 发明授权

US07277538B2 Distortion compensation in an acoustic echo canceler 有权
标题翻译：声回波消除器中的失真补偿
公开(公告)号：US07277538B2
公开(公告)日：2007-10-02
申请号：US11243236
申请日：2005-10-04
申请人： James H. Parry
发明人： James H. Parry
IPC分类号： H04M9/08
CPC分类号： H04M9/082
摘要： An audio communications system has an acoustic echo cancellation (AEC) module. The AEC module receives a digital signal sent to a loudspeaker and a digital signal received from a microphone. The signal received from the microphone contains an echo of the signal played through the loudspeaker. The loudspeaker signal is processed by an audio generation module (AGM) that models substantially nonlinear distortions that can occur while producing the signal played through the loudspeaker. The AGM includes a modeling path comprised of one or more distortion modules. Each distortion module receives digital samples as input, modifies the samples to model a form of distortion, and outputs the modified samples. The output of the AGM is provided to an acoustic echo estimation (AEE) module that uses adaptive algorithms to compensate for substantially linear changes in the echo characteristics of the environment in which the loudspeaker and microphone are located.
摘要翻译：音频通信系统具有声学回声消除（AEC）模块。 AEC模块接收发送到扬声器的数字信号和从麦克风接收的数字信号。从麦克风接收的信号包含通过扬声器播放的信号的回波。扬声器信号由音频产生模块（AGM）处理，该模块（AGM）在产生通过扬声器播放的信号的同时产生基本上非线性失真。 AGM包括由一个或多个失真模块组成的建模路径。每个失真模块接收数字样本作为输入，修改样本以建模失真形式，并输出修改后的样本。 AGM的输出被提供给声学回声估计（AEE）模块，其使用自适应算法来补偿扬声器和麦克风位于其中的环境的回波特性中的基本线性变化。

7. 发明授权

US06603501B1 Videoconferencing using distributed processing 有权
标题翻译：视频会议使用分布式处理
公开(公告)号：US06603501B1
公开(公告)日：2003-08-05
申请号：US09721547
申请日：2000-11-21
申请人： James H. Parry , See-Mong Tan
发明人： James H. Parry , See-Mong Tan
IPC分类号： H04N714
CPC分类号： H04N7/152 , H04N7/147
摘要： A video teleconferencing system and method transfers video teleconferencing signals from a sender to a receiver. The sender determines decision information based on internal or external factors. The sender may or may not generate a video teleconferencing signal depending on the content of the decision information. If generated, the video teleconferencing signal is encoded at the sender and sent to the receiver. Each sender includes at least one memory module for storing the decoded signal, each memory module is one group. The sender updates its memory module with a copy of each sent video teleconferencing signal. The receiver decodes the signal and presents the signal to the user of the receiver. The receiver stores a copy of the signal in a memory module identified with each specific sender.
摘要翻译：视频电话会议系统和方法将视频电话会议信号从发送方传送到接收方。发件人根据内部或外部因素确定决策信息。根据决策信息的内容，发送者可以产生或不产生视频电话会议信号。如果生成，则视频电话会议信号在发送器处被编码并发送到接收器。每个发送器包括用于存储解码信号的至少一个存储器模块，每个存储器模块是一组。发送者使用每个发送的视频电话会议信号的副本更新其内存模块。接收机对信号进行解码并将信号呈现给接收机的用户。接收器将信号的副本存储在由每个特定发送者识别的存储器模块中。

8. 发明授权

US08422663B1 Structure and method for echo reduction without loss of information 有权
标题翻译：回波减少的结构和方法，而不会丢失信息
公开(公告)号：US08422663B1
公开(公告)日：2013-04-16
申请号：US13168762
申请日：2011-06-24
申请人： James H. Parry
发明人： James H. Parry
IPC分类号： H04M9/08 , H04M1/00 , H04B1/38 , H04B3/20 , H04L5/16
CPC分类号： H04M9/082 , G10L25/87 , H04M1/6016
摘要： An echo reduction method stores a received audio information stream. A sound detection flag is activated following detection of locally generated sound. Output based on the received audio information stream is muted in response to the activating the sound detection flag. Rendering status of the received audio information stream is saved, in response to the activating the sound detection flag, to reduce loss of audio information. At least a portion of the stored received audio information stream is rendered following inactivation of the sound detection flag.
摘要翻译：回波减少方法存储所接收的音频信息流。检测到本地产生的声音后，声音检测标志被激活。响应于激活声音检测标志，基于接收到的音频信息流的输出被静音。响应于激活声音检测标志，节省了所接收的音频信息流的渲染状态，以减少音频信息的丢失。存储的所接收的音频信息流的至少一部分在声音检测标志失效之后呈现。

9. 发明授权

US07995745B1 Structure and method for echo reduction without loss of information 有权
标题翻译：回波减少的结构和方法，而不会丢失信息
公开(公告)号：US07995745B1
公开(公告)日：2011-08-09
申请号：US11502809
申请日：2006-08-11
申请人： James H. Parry
发明人： James H. Parry
IPC分类号： H04M9/08
CPC分类号： H04M9/082 , G10L25/87 , H04M1/6016
摘要： An echo reduction method stores a received audio information stream. A sound detection flag is activated following detection of locally generated sound. Output based on the received audio information stream is muted in response to the activating the sound detection flag. Rendering status of the received audio information stream is saved, in response to the activating the sound detection flag, to reduce loss of audio information. At least a portion of the stored received audio information stream is rendered following inactivation of the sound detection flag.
摘要翻译：回波减少方法存储所接收的音频信息流。检测到本地产生的声音后，声音检测标志被激活。响应于激活声音检测标志，基于所接收的音频信息流的输出被静音。响应于激活声音检测标志，节省了所接收的音频信息流的渲染状态，以减少音频信息的丢失。存储的所接收的音频信息流的至少一部分在声音检测标志失效之后呈现。

10. 发明授权

US06728312B1 Adaptive video decoding and rendering with respect to processor congestion 有权
标题翻译：相对于处理器拥塞的自适应视频解码和渲染
公开(公告)号：US06728312B1
公开(公告)日：2004-04-27
申请号：US09548570
申请日：2000-04-13
申请人： Paul H. Whitford , James H. Parry , See-Mong Tan
发明人： Paul H. Whitford , James H. Parry , See-Mong Tan
IPC分类号： H04N712
CPC分类号： H04N19/156 , H04N19/12
摘要： Disclosed is a method and system for reducing audio artifacts and/or avoiding invalid reference memory in a compressed video decoder due to processor congestion. The system and method includes decoding compressed frames of video data, and decoding compressed frames of audio data. The system and method determines whether audio data can be generated without audio artifacts. If it is determined that audio can be generated without audio artifacts, then images corresponding to the decoded frames of video data, respectively, are subsequently displayed. If however it is determined that the audio can not be generated without artifacts due to processor congestion, images are displayed and redisplayed while processor power is shifted to decoding frames of audio data.
摘要翻译：公开了一种用于在压缩视频解码器中由于处理器拥塞而减少音频伪影和/或避免无效参考存储器的方法和系统。该系统和方法包括解码视频数据的压缩帧，以及解码压缩的音频数据帧。系统和方法确定是否可以生成音频数据而不产生音频伪影。如果确定可以生成没有音频伪影的音频，则随后显示与解码的视频数据帧相对应的图像。然而，如果确定由于处理器拥塞而无法产生音频不产生伪像，则在处理器功率转换到解码帧音频数据时，显示和重新显示图像。

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