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。

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。

公开(公告)号：US20080069127A1

公开(公告)日：2008-03-20

申请号：US11522268

申请日：2006-09-15

申请人： Hosam A. Khalil ,  Guo-Wei Shieh ,  Tian Wang

发明人： Hosam A. Khalil ,  Guo-Wei Shieh ,  Tian Wang

IPC分类号： H04L12/56 ,  H04J3/06

CPC分类号： H04J3/062 ,  H04J3/0626 ,  H04J3/0632 ,  H04L47/283 ,  H04L47/30

摘要： A method of compensating for jitter in a packet stream is described. The method comprises placing undecoded frames extracted from packets in the packet stream into a jitter buffer while decoding frames from the jitter buffer and placing the decoded frames into a sample buffer at a rate determined using an average playout delay. The average playout delay is the running average of the playout delay calculated for each packet as each packet becomes available. The playout delay for each packet is the sum of a sample buffer delay and a jitter buffer delay. As each packet is received, the average playout delay is adjusted based on a comparison of the playout delay associated with the received packet to the current average playout delay.

摘要翻译： 描述了补偿分组流中的抖动的方法。 该方法包括将从分组流中的分组提取的未解码的帧放入抖动缓冲器中，同时从抖动缓冲器解码帧并将解码的帧以使用平均播出延迟确定的速率放置到采样缓冲器中。 平均播出延迟是每个分组变得可用时为每个分组计算的播出延迟的运行平均值。 每个数据包的播出延迟是采样缓冲区延迟和抖动缓冲区延迟的总和。 当接收到每个分组时，基于与接收分组相关联的播放延迟与当前平均播放延迟的比较来调整平均播出延迟。

公开(公告)号：US20080059161A1

公开(公告)日：2008-03-06

申请号：US11470577

申请日：2006-09-06

申请人： Hosam A. Khalil ,  Tian Wang

发明人： Hosam A. Khalil ,  Tian Wang

IPC分类号： G10L21/02

CPC分类号： G10L19/012

摘要： This document describes tools capable of enabling and/or adaptively generating comfort noise. The tools may do so by receiving some background noise, analyzing that noise, and generating comfort noise based on the received background noise. In some embodiments, for example, the tools build and continuously adapt a history based on segments of background noise as they are received from the sender. The tools may use this history to generate comfort noise that is pleasing, relatively accurate, and/or dynamically changing responsive to changes in a speaker's background noise.

摘要翻译： 本文档描述了能够启用和/或自适应地产生舒适噪声的工具。 这些工具可以通过接收一些背景噪声，分析噪声和基于接收的背景噪声产生舒适噪声来实现。 在一些实施例中，例如，当它们从发送者接收时，工具构建并连续地适应基于背景噪声的段的历史。 这些工具可以使用这种历史来产生响应于扬声器背景噪声的变化而令人愉快，相对准确和/或动态改变的舒适噪声。

公开(公告)号：US08005670B2

公开(公告)日：2011-08-23

申请号：US11873707

申请日：2007-10-17

申请人： Hosam A. Khalil ,  Guo-Wei Shieh

发明人： Hosam A. Khalil ,  Guo-Wei Shieh

IPC分类号： G10L21/02 ,  G10L19/00

CPC分类号： G10L19/005

摘要： To reduce audio glitch rendering buffer of an audio application is pre-filled with natural sounding audio rather than zeros. For every frame of audio sent for rendering, the rendering buffer is also pre-filled or the signal is stretched in the buffer in anticipation of a glitch. If the glitch does not occur, then the stretched signal is overwritten and the end user does not notice it. If the glitch does occur, then the rendering buffer is already filled with a stretched version of the previous audio and may result in sound that is acceptable. After recovery from the glitch, any new data is smoothly merged into the fake audio that was generated before.

摘要翻译： 为了减少音频应用程序的音频毛刺渲染缓冲区，预先填充了自然的声音音频而不是零。 对于发送用于渲染的每个音频帧，渲染缓冲区也是预填充的，或者信号在缓冲区中被延伸以预期出现故障。 如果没有发生毛刺，则被拉伸的信号被覆盖，最终用户没有注意到。 如果发生毛刺，则渲染缓冲区已经填充了先前音频的拉伸版本，并且可能导致可接受的声音。 从故障恢复后，任何新的数据都会平滑地合并到之前生成的假音频中。

公开(公告)号：US06941263B2

公开(公告)日：2005-09-06

申请号：US09896062

申请日：2001-06-29

申请人： Hong Wang ,  Vladimir Cuperman ,  Allen Gersho ,  Hosam A. Khalil

发明人： Hong Wang ,  Vladimir Cuperman ,  Allen Gersho ,  Hosam A. Khalil

IPC分类号： G10L19/02 ,  G10L11/00 ,  G10L19/14 ,  G10L21/02 ,  H03M7/30

CPC分类号： G10L19/26 ,  G10L21/0364

摘要： A method and system of performing postfiltering in the frequency domain to improve the quality of a speech signal, especially for synthesized speech resulting from codecs of low bit-rate, is provided. The method comprises LPC tilt computation and compensation methods and modules, a formant filter gain computation method and module, and an anti-aliasing method and module. The formant filter gain calculation employs an LPC representation, an all-pole modeling, a non-linear transformation and a phase computation. The LPC used for deriving the postfilter may be transmitted from an encoder or may be estimated from a synthesized or other speech signal in a decoder or receiver. The invention may be implemented in a linked decoder and encoder. A separate LPC evaluation unit that is responsible for processing and or deriving the LPC may be implemented within the invention.

摘要翻译： 提供了一种在频域中执行后置滤波以提高语音信号质量的方法和系统，特别是对于由低比特率的编解码器产生的合成语音。 该方法包括LPC倾斜计算和补偿方法和模块，共振峰滤波器增益计算方法和模块，以及抗混叠方法和模块。 共振峰滤波器增益计算采用LPC表示，全极建模，非线性变换和相位计算。 用于导出后置滤波器的LPC可以从编码器发送，或者可以从解码器或接收机中的合成的或其他语音信号来估计。 本发明可以在链接的解码器和编码器中实现。 负责处理和推导LPC的单独的LPC评估单元可以在本发明中实现。

公开(公告)号：US20090106020A1

公开(公告)日：2009-04-23

申请号：US11873707

申请日：2007-10-17

申请人： Hosam A. Khalil ,  Guo-Wei Shieh

发明人： Hosam A. Khalil ,  Guo-Wei Shieh

IPC分类号： G10L11/02

CPC分类号： G10L19/005

摘要： To reduce audio glitch rendering buffer of an audio application is pre-filled with natural sounding audio rather than zeros. For every frame of audio sent for rendering, the rendering buffer is also pre-filled or the signal is stretched in the buffer in anticipation of a glitch. If the glitch does not occur, then the stretched signal is overwritten and the end user does not notice it. If the glitch does occur, then the rendering buffer is already filled with a stretched version of the previous audio and may result in sound that is acceptable. After recovery from the glitch, any new data is smoothly merged into the fake audio that was generated before.

摘要翻译： 为了减少音频应用程序的音频毛刺渲染缓冲区，预先填充了自然的声音音频而不是零。 对于发送用于渲染的每个音频帧，渲染缓冲区也是预填充的，或者信号在缓冲区中被延伸以预期出现故障。 如果没有发生毛刺，则被拉伸的信号被覆盖，最终用户没有注意到。 如果发生毛刺，则渲染缓冲区已经填充了先前音频的拉伸版本，并且可能导致可接受的声音。 从故障恢复后，任何新的数据都会平滑地合并到之前生成的假音频中。

公开(公告)号：US08379800B2

公开(公告)日：2013-02-19

申请号：US13075130

申请日：2011-03-29

申请人： Warren M. Lam ,  Hosam A. Khalil ,  Vinod Prakash ,  Sang K. Choe ,  Hong Wang Sodoma

发明人： Warren M. Lam ,  Hosam A. Khalil ,  Vinod Prakash ,  Sang K. Choe ,  Hong Wang Sodoma

IPC分类号： H04M3/08

CPC分类号： H04M3/56 ,  H04M3/2227

摘要： Detecting at least one of an echo detector and a noise detector based on analysis of audio streams transmitted to and received from each endpoint of a conference. When certain characteristics of the respective audio streams for a given endpoint are classified as significant against certain criteria, a determination is made as to whether that endpoint is a source of echo and/or noise. Subsequent actions are taken to alert users and/or prevent broadcast of impaired signals.

摘要翻译： 基于对会议的每个端点发送和接收的音频流的分析来检测回波检测器和噪声检测器中的至少一个。 当给定端点的相应音频流的某些特性被分类为对某些标准是显着的时，确定该端点是否是回波源和/或噪声源。 采取后续行动来提醒用户和/或防止受损信号的广播。

公开(公告)号：US20120250830A1

公开(公告)日：2012-10-04

申请号：US13075130

申请日：2011-03-29

申请人： Warren M. Lam ,  Hosam A. Khalil ,  Vinod Prakash ,  Sang K. Choe ,  Hong Wang Sodoma

发明人： Warren M. Lam ,  Hosam A. Khalil ,  Vinod Prakash ,  Sang K. Choe ,  Hong Wang Sodoma

IPC分类号： H04M3/08

CPC分类号： H04M3/56 ,  H04M3/2227

摘要： Detecting at least one of an echo detector and a noise detector based on analysis of audio streams transmitted to and received from each endpoint of a conference. When certain characteristics of the respective audio streams for a given endpoint are classified as significant against certain criteria, a determination is made as to whether that endpoint is a source of echo and/or noise. Subsequent actions are taken to alert users and/or prevent broadcast of impaired signals.

摘要翻译： 基于对会议的每个端点发送和接收的音频流的分析来检测回波检测器和噪声检测器中的至少一个。 当给定端点的相应音频流的某些特性被分类为对某些标准是显着的时，确定该端点是否是回波源和/或噪声源。 采取后续行动来提醒用户和/或防止受损信号的广播。

公开(公告)号：US07124077B2

公开(公告)日：2006-10-17

申请号：US11045907

申请日：2005-01-28

申请人： Hong Wang ,  Vladimir Cuperman ,  Allen Gersho ,  Hosam A. Khalil

发明人： Hong Wang ,  Vladimir Cuperman ,  Allen Gersho ,  Hosam A. Khalil

IPC分类号： G10L19/04 ,  G10L21/02

CPC分类号： G10L19/26 ,  G10L21/0364

摘要： A method and system of performing postfiltering in the frequency domain to improve the quality of a speech signal, especially for synthesized speech resulting from codecs of low bit-rate, is provided. The method comprises LPC tilt computation and compensation methods and modules, a formant filter gain computation method and module, and an anti-aliasing method and module. The formant filter gain calculation employs an LPC representation, an all-pole modeling, a non-linear transformation and a phase computation. The LPC used for deriving the postfilter may be transmitted from an encoder or may be estimated from a synthesized or other speech signal in a decoder or receiver. The invention may be implemented in a linked decoder and encoder. A separate LPC evaluation unit that is responsible for processing and or deriving the LPC may be implemented within the invention.