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1. 发明授权

US09047865B2 Scalable and embedded codec for speech and audio signals 有权
标题翻译：用于语音和音频信号的可扩展和嵌入式编解码器
公开(公告)号：US09047865B2
公开(公告)日：2015-06-02
申请号：US11889332
申请日：2007-08-10
申请人： Joseph Gerard Aguilar , David A. Campana , Juin-Hwey Chen , Robert B. Dunn , Robert J. McAulay , Xiaoquin Sun , Wei Wang , Craig Watkins , Robert W. Zopf
发明人： Joseph Gerard Aguilar , David A. Campana , Juin-Hwey Chen , Robert B. Dunn , Robert J. McAulay , Xiaoquin Sun , Wei Wang , Craig Watkins , Robert W. Zopf
IPC分类号： G10L19/18 , G10L19/093 , G10L19/24
CPC分类号： G10L19/002 , G10L19/093 , G10L19/24
摘要： A system and method for processing of audio and speech signals is disclosed, which provide compatibility over a range of communication devices operating at different sampling frequencies and/or bit rates. The analyzer of the system divides the input signal in different portions, at least one of which carries information sufficient to provide intelligible reconstruction of the input signal. The analyzer also encodes separate information about other portions of the signal in an embedded manner, so that a smooth transition can be achieved from low bit-rate to high bit-rate applications. Accordingly, communication devices operating at different sampling rates and/or bit-rates can extract corresponding information from the output bit stream of the analyzer. In the present invention embedded information generally relates to separate parameters of the input signal, or to additional resolution in the transmission of original signal parameters. Non-linear techniques for enhancing the overall performance of the system are also disclosed. Also disclosed is a novel method of improving the quantization of signal parameters. In a specific embodiment the input signal is processed in two or more modes dependent on the state of the signal in a frame. When the signal is determined to be in a transition state, the encoder provides phase information about N sinusoids, which the decoder end uses to improve the quality of the output signal at low bit rates.
摘要翻译：公开了一种用于处理音频和语音信号的系统和方法，其提供了在不同采样频率和/或比特率下操作的通信设备的范围上的兼容性。系统的分析器将输入信号分成不同的部分，其中至少一个传送足以提供输入信号的可理解的重建的信息。分析器还以嵌入的方式编码关于信号的其他部分的单独信息，从而可以从低比特率到高比特率应用实现平滑的转换。因此，以不同的采样率和/或比特率工作的通信设备可以从分析器的输出比特流中提取相应的信息。在本发明中，嵌入信息通常涉及输入信号的单独参数，或者涉及原始信号参数传输中的附加分辨率。还公开了用于增强系统的整体性能的非线性技术。还公开了一种改进信号参数量化的新方法。在具体实施例中，输入信号以两个或更多个模式被处理，这取决于帧中的信号的状态。当信号被确定为处于转换状态时，编码器提供关于N个正弦曲线的相位信息，解码器端用于以低比特率提高输出信号的质量。

2. 发明授权

US07257535B2 Parametric speech codec for representing synthetic speech in the presence of background noise 有权
标题翻译：用于在背景噪声的存在下表示合成语音的参数语音编解码器
公开(公告)号：US07257535B2
公开(公告)日：2007-08-14
申请号：US11261969
申请日：2005-10-28
申请人： Joseph Gerard Aguilar , Juin-Hwey Chen , Wei Wang , Robert W. Zopf
发明人： Joseph Gerard Aguilar , Juin-Hwey Chen , Wei Wang , Robert W. Zopf
IPC分类号： G10L13/02
CPC分类号： G10L19/265 , G10L19/093 , G10L21/0272 , G10L25/18 , G10L25/30 , G10L25/90 , G10L25/93
摘要： A system and method are provided for processing audio and speech signals using a pitch and voicing dependent spectral estimation algorithm (voicing algorithm) to accurately represent voiced speech, unvoiced speech, and mixed speech in the presence of background noise, and background noise with a single model. The present invention also modifies the synthesis model based on an estimate of the current input signal to improve the perceptual quality of the speech and background noise under a variety of input conditions. The present invention also improves the voicing dependent spectral estimation algorithm robustness by introducing the use of a Multi-Layer Neural Network in the estimation process. The voicing dependent spectral estimation algorithm provides an accurate and robust estimate of the voicing probability under a variety of background noise conditions. This is essential to providing high quality intelligible speech in the presence of background noise.
摘要翻译：提供了一种用于处理音频和语音信号的系统和方法，该音频和语音信号使用音调和语音相关频谱估计算法（语音算法）来在背景噪声的存在下准确地表示浊音，清音语音和混合语音，并且使用单个模型。本发明还基于当前输入信号的估计来修改合成模型，以在各种输入条件下提高语音的感知质量和背景噪声。本发明还通过在估计过程中引入多层神经网络的使用来改进语音依赖频谱估计算法的鲁棒性。语音依赖频谱估计算法在各种背景噪声条件下提供对发音概率的准确和鲁棒的估计。这对于在背景噪声的存在下提供高质量的可理解的语音是至关重要的。

3. 发明授权

US07092881B1 Parametric speech codec for representing synthetic speech in the presence of background noise 有权
标题翻译：用于在背景噪声的存在下表示合成语音的参数语音编解码器
公开(公告)号：US07092881B1
公开(公告)日：2006-08-15
申请号：US09625960
申请日：2000-07-26
申请人： Joseph Gerard Aguilar , Juin-Hwey Chen , Wei Wang , Robert W. Zopf
发明人： Joseph Gerard Aguilar , Juin-Hwey Chen , Wei Wang , Robert W. Zopf
IPC分类号： G10L15/20
CPC分类号： G10L19/265 , G10L19/093 , G10L21/0272 , G10L25/18 , G10L25/30 , G10L25/90 , G10L25/93
摘要： A system and method are provided for processing audio and speech signals using a pitch and voicing dependent spectral estimation algorithm (voicing algorithm) to accurately represent voiced speech, unvoiced speech, and mixed speech in the presence of background noise, and background noise with a single model. The present invention also modifies the synthesis model based on an estimate of the current input signal to improve the perceptual quality of the speech and background noise under a variety of input conditions. The present invention also improves the voicing dependent spectral estimation algorithm robustness by introducing the use of a Multi-Layer Neural Network in the estimation process. The voicing dependent spectral estimation algorithm provides an accurate and robust estimate of the voicing probability under a variety of background noise conditions. This is essential to providing high quality intelligible speech in the presence of background noise.
摘要翻译：提供了一种用于处理音频和语音信号的系统和方法，该音频和语音信号使用音调和语音相关频谱估计算法（语音算法）来在背景噪声的存在下准确地表示浊音，清音语音和混合语音，并且使用单个模型。本发明还基于当前输入信号的估计来修改合成模型，以在各种输入条件下提高语音的感知质量和背景噪声。本发明还通过在估计过程中引入多层神经网络的使用来改进语音依赖频谱估计算法的鲁棒性。语音依赖频谱估计算法在各种背景噪声条件下提供对发音概率的准确和鲁棒的估计。这对于在背景噪声的存在下提供高质量的可理解的语音是至关重要的。

4. 发明授权

US07272556B1 Scalable and embedded codec for speech and audio signals 有权
标题翻译：用于语音和音频信号的可扩展和嵌入式编解码器
公开(公告)号：US07272556B1
公开(公告)日：2007-09-18
申请号：US09159481
申请日：1998-09-23
申请人： Joseph Gerard Aguilar , David A. Campana , Raymond Chen , Robert B. Dunn , Robert J. McAulay , Xiaoquin Sun , Wei Wang , Craig Watkins , Robert W. Zopf
发明人： Joseph Gerard Aguilar , David A. Campana , Raymond Chen , Robert B. Dunn , Robert J. McAulay , Xiaoquin Sun , Wei Wang , Craig Watkins , Robert W. Zopf
IPC分类号： G10L21/00
CPC分类号： G10L19/002 , G10L19/093 , G10L19/24
摘要： A system and method for processing of audio and speech signals is disclosed, which provide compatibility over a range of communication devices operating at different sampling frequencies and/or bit rates. The analyzer of the system divides the input signal in different portions, at least one of which carries information sufficient to provide intelligible reconstruction of the input signal. The analyzer also encodes separate information about other portions of the signal in an embedded manner, so that a smooth transition can be achieved from low bit-rate to high bit-rate applications. Accordingly, communication devices operating at different sampling rates and/or bit-rates can extract corresponding information from the output bit stream of the analyzer. In the present invention embedded information generally relates to separate parameters of the input signal, or to additional resolution in the transmission of original signal parameters. Non-linear techniques for enhancing the overall performance of the system are also disclosed. Also disclosed is a novel method of improving the quantization of signal parameters. In a specific embodiment the input signal is processed in two or more modes dependent on the state of the signal in a frame. When the signal is determined to be in a transition state, the encoder provides phase information about N sinusoids, which the decoder end uses to improve the quality of the output signal at low bit rates.
摘要翻译：公开了一种用于处理音频和语音信号的系统和方法，其提供了在不同采样频率和/或比特率下操作的通信设备的范围上的兼容性。系统的分析器将输入信号分成不同的部分，其中至少一个传送足以提供输入信号的可理解的重建的信息。分析器还以嵌入的方式编码关于信号的其他部分的单独信息，从而可以从低比特率到高比特率应用实现平滑的转换。因此，以不同的采样率和/或比特率工作的通信设备可以从分析器的输出比特流中提取相应的信息。在本发明中，嵌入信息通常涉及输入信号的单独参数，或者涉及原始信号参数传输中的附加分辨率。还公开了用于增强系统的整体性能的非线性技术。还公开了一种改进信号参数量化的新方法。在具体实施例中，输入信号以两个或更多个模式被处理，这取决于帧中的信号的状态。当信号被确定为处于转换状态时，编码器提供关于N个正弦曲线的相位信息，解码器端用于以低比特率提高输出信号的质量。

5. 发明授权

US08670990B2 Dynamic time scale modification for reduced bit rate audio coding 有权
标题翻译：用于降低比特率音频编码的动态时间尺度修改
公开(公告)号：US08670990B2
公开(公告)日：2014-03-11
申请号：US12847120
申请日：2010-07-30
申请人： Juin-Hwey Chen , Hong-Goo Kang , Robert W. Zopf , Jes Thyssen
发明人： Juin-Hwey Chen , Hong-Goo Kang , Robert W. Zopf , Jes Thyssen
IPC分类号： G10L21/04 , G10L11/06
CPC分类号： G10L19/22 , G10L19/08
摘要： Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder.
摘要翻译：描述了利用动态时间尺度修正（TSM）来实现降低的比特率音频编码的系统和方法。根据实施例，在由编码器对TSM压缩进行编码之前，将不同级别的TSM压缩选择性地应用于输入语音信号的段。编码的TSM压缩段在解码器处被接收，解码器对这些段进行解码，然后基于从编码器接收的信息向每个TSM解压缩应用适当级别的TSM解压缩。通过在编码之前选择性地对输入语音信号的段应用不同级别的TSM压缩，减少与编码器/解码器相关联的编码比特率。此外，通过选择考虑到该信号的某些局部特性的输入语音信号的每个段的TSM压缩级别，提供这样的比特率降低，而不会将不可接受的失真电平引入到由解码器产生的输出语音信号中。

6. 发明申请

US20110029317A1 DYNAMIC TIME SCALE MODIFICATION FOR REDUCED BIT RATE AUDIO CODING 有权
标题翻译：用于减少比特率音频编码的动态时间尺度修改
公开(公告)号：US20110029317A1
公开(公告)日：2011-02-03
申请号：US12847120
申请日：2010-07-30
申请人： Juin-Hwey Chen , Hong-goo Kang , Robert W. Zopf , Jes Thyssen
发明人： Juin-Hwey Chen , Hong-goo Kang , Robert W. Zopf , Jes Thyssen
IPC分类号： G10L19/00
CPC分类号： G10L19/22 , G10L19/08
摘要： Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder.
摘要翻译：描述了利用动态时间尺度修正（TSM）来实现降低的比特率音频编码的系统和方法。根据实施例，在由编码器对TSM压缩进行编码之前，将不同级别的TSM压缩选择性地应用于输入语音信号的段。编码的TSM压缩段在解码器处被接收，解码器对这些段进行解码，然后基于从编码器接收的信息向每个TSM解压缩应用适当级别的TSM解压缩。通过在编码之前选择性地对输入语音信号的段应用不同级别的TSM压缩，减少与编码器/解码器相关联的编码比特率。此外，通过选择考虑到该信号的某些局部特性的输入语音信号的每个段的TSM压缩级别，提供这样的比特率降低，而不会将不可接受的失真电平引入到由解码器产生的输出语音信号中。

7. 发明授权

US08000960B2 Packet loss concealment for sub-band predictive coding based on extrapolation of sub-band audio waveforms 有权
标题翻译：基于子带音频波形外推的子带预测编码的分组丢失隐藏
公开(公告)号：US08000960B2
公开(公告)日：2011-08-16
申请号：US11838891
申请日：2007-08-15
申请人： Juin-Hwey Chen , Robert W. Zopf , Jes Thyssen
发明人： Juin-Hwey Chen , Robert W. Zopf , Jes Thyssen
IPC分类号： G10L19/10
CPC分类号： G10L19/0204 , G10L19/005 , G10L19/04
摘要： A technique is described for concealing the effect of a lost frame in a series of frames representing an encoded audio signal in a sub-band predictive coding system. In accordance with the technique, a first synthesized sub-band audio signal is synthesized, wherein synthesizing the first synthesized sub-band audio signal comprises performing waveform extrapolation based on a stored first sub-band decoded audio signal. A second synthesized sub-band audio signal is also synthesized, wherein synthesizing the second synthesized sub-band audio signal comprises performing waveform extrapolation based on the stored second sub-band decoded audio signal. The first synthesized sub-band audio signal and the second synthesized sub-band audio signal are combined to generate a synthesized full-band output audio signal corresponding to a lost frame.
摘要翻译：描述了一种用于在子带预测编码系统中隐藏表示编码音频信号的一系列帧中的丢失帧的影响的技术。根据该技术，合成第一合成子带音频信号，其中合成第一合成子带音频信号包括基于存储的第一子带解码音频信号执行波形外推。还合成了第二合成子带音频信号，其中合成第二合成子带音频信号包括基于所存储的第二子带解码音频信号执行波形外推。第一合成子带音频信号和第二合成子带音频信号被组合以产生对应于丢失帧的合成全频带输出音频信号。

8. 发明申请

US20090281797A1 BIT ERROR CONCEALMENT FOR AUDIO CODING SYSTEMS 有权
标题翻译：音频编码系统的位错误保存
公开(公告)号：US20090281797A1
公开(公告)日：2009-11-12
申请号：US12431155
申请日：2009-04-28
申请人： Robert W. Zopf , Vivek Kumar , Juin-Hwey Chen
发明人： Robert W. Zopf , Vivek Kumar , Juin-Hwey Chen
IPC分类号： G10L19/00 , G10L11/04 , G10L21/02
CPC分类号： G10L19/005
摘要： A bit error concealment (BEC) system and method is described herein that detects and conceals the presence of click-like artifacts in an audio signal caused by bit errors introduced during transmission of the audio signal within an audio communications system. A particular embodiment of the present invention utilizes a low-complexity design that introduces no added delay and that is particularly well-suited for applications such as Bluetooth® wireless audio devices which have low cost and low power dissipation requirements.
摘要翻译：本文描述了一种错误隐藏（BEC）系统和方法，其检测和隐藏由音频通信系统内的音频信号传输期间引入的位错误引起的音频信号中出现的点击样伪像。本发明的一个具体实施例利用了不增加延迟的低复杂度设计，特别适用于具有低成本和低功耗要求的蓝牙无线音频设备的应用。

9. 发明申请

US20090248405A1 PACKET LOSS CONCEALMENT FOR A SUB-BAND PREDICTIVE CODER BASED ON EXTRAPOLATION OF EXCITATION WAVEFORM 有权
标题翻译：基于激励波形扩展的子带预测编码器的分组丢失隐藏
公开(公告)号：US20090248405A1
公开(公告)日：2009-10-01
申请号：US12474809
申请日：2009-05-29
申请人： Juin-Hwey Chen , Jes Thyssen , Robert W. Zopf
发明人： Juin-Hwey Chen , Jes Thyssen , Robert W. Zopf
IPC分类号： G10L19/00
CPC分类号： G10L19/0208 , G10L19/005
摘要： Systems and methods are described for performing packet loss concealment using an extrapolation of an excitation waveform in a sub-band predictive speech coder, such as an ITU-T Recommendation G.722 wideband speech coder. The systems and methods are useful for concealing the quality-degrading effects of packet loss in a sub-band predictive coder and address some sub-band architectural issues when applying excitation extrapolation techniques to such sub-band predictive coders.
摘要翻译：描述了使用在诸如ITU-T G.722建议书G.722宽带语音编码器的子带预测语音编码器中外推激励波形来执行分组丢失隐藏的系统和方法。这些系统和方法对于隐藏子带预测编码器中的分组丢失的质量降级效应是有用的，并且当向这种子带预测编码器应用激励外推技术时，解决某些子带架构问题。

10. 发明申请

US20080033718A1 Classification-Based Frame Loss Concealment for Audio Signals 有权
标题翻译：音频信号基于分类的帧丢失隐藏
公开(公告)号：US20080033718A1
公开(公告)日：2008-02-07
申请号：US11734800
申请日：2007-04-13
申请人： Robert W. Zopf , Juin-Hwey Chen , Jes Thyssen
发明人： Robert W. Zopf , Juin-Hwey Chen , Jes Thyssen
IPC分类号： G10L19/02
CPC分类号： G10L19/005 , G10L25/78
摘要： An audio decoding system performs frame loss concealment (FLC) when portions of a bit stream representing an audio signal are lost within the context of a digital communication system. The audio decoding system employs two different FLC methods: one designed to perform well for music, and the other designed to perform well for speech. When a frame is deemed lost, the audio decoding system analyzes a previously-decoded audio signal corresponding to previously-decoded frames of an audio bit-stream. Based on the results of the analysis, the lost frame is classified as either speech or music. Using this classification, other signal analysis, and knowledge of the employed FLC methods, the audio decoding system selects the appropriate FLC method which then performs FLC on the lost frame.
摘要翻译：音频解码系统在数字通信系统的上下文中丢失表示音频信号的比特流的部分时执行帧丢失隐藏（FLC）。音频解码系统采用两种不同的FLC方法：一种被设计为对音乐表现良好，另一种被设计为对演讲表现良好。当帧被认为丢失时，音频解码系统分析对应于音频比特流的先前解码的帧的先前解码的音频信号。基于分析结果，丢失的帧被分类为语音或音乐。使用这种分类，其他信号分析和所采用的FLC方法的知识，音频解码系统选择适当的FLC方法，然后在丢帧上执行FLC。

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