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    • 1. 发明授权
    • Scalable and embedded codec for speech and audio signals
    • 用于语音和音频信号的可扩展和嵌入式编解码器
    • US07272556B1
    • 2007-09-18
    • US09159481
    • 1998-09-23
    • Joseph Gerard AguilarDavid A. CampanaRaymond ChenRobert B. DunnRobert J. McAulayXiaoquin SunWei WangCraig WatkinsRobert W. Zopf
    • Joseph Gerard AguilarDavid A. CampanaRaymond ChenRobert B. DunnRobert J. McAulayXiaoquin SunWei WangCraig WatkinsRobert W. Zopf
    • G10L21/00
    • G10L19/002G10L19/093G10L19/24
    • 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.
    • 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
    • Scalable and embedded codec for speech and audio signals
    • 用于语音和音频信号的可扩展和嵌入式编解码器
    • US09047865B2
    • 2015-06-02
    • US11889332
    • 2007-08-10
    • Joseph Gerard AguilarDavid A. CampanaJuin-Hwey ChenRobert B. DunnRobert J. McAulayXiaoquin SunWei WangCraig WatkinsRobert W. Zopf
    • Joseph Gerard AguilarDavid A. CampanaJuin-Hwey ChenRobert B. DunnRobert J. McAulayXiaoquin SunWei WangCraig WatkinsRobert W. Zopf
    • G10L19/18G10L19/093G10L19/24
    • G10L19/002G10L19/093G10L19/24
    • 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.
    • 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。
    • 3. 发明申请
    • Scalable and embedded codec for speech and audio signals
    • 用于语音和音频信号的可扩展和嵌入式编解码器
    • US20080052068A1
    • 2008-02-28
    • US11889332
    • 2007-08-10
    • Joseph AguilarDavid CampanaJuin-Hwey (Raymond) ChenRobert DunnRobert McAulayXiaoquin SunWei WangCraig WatkinsRobert Zopf
    • Joseph AguilarDavid CampanaJuin-Hwey (Raymond) ChenRobert DunnRobert McAulayXiaoquin SunWei WangCraig WatkinsRobert Zopf
    • G10L21/00
    • G10L19/002G10L19/093G10L19/24
    • 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.
    • 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。
    • 4. 发明授权
    • Parametric speech codec for representing synthetic speech in the presence of background noise
    • 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
    • US07257535B2
    • 2007-08-14
    • US11261969
    • 2005-10-28
    • Joseph Gerard AguilarJuin-Hwey ChenWei WangRobert W. Zopf
    • Joseph Gerard AguilarJuin-Hwey ChenWei WangRobert W. Zopf
    • G10L13/02
    • G10L19/265G10L19/093G10L21/0272G10L25/18G10L25/30G10L25/90G10L25/93
    • 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.
    • 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。
    • 5. 发明授权
    • Parametric speech codec for representing synthetic speech in the presence of background noise
    • 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
    • US07092881B1
    • 2006-08-15
    • US09625960
    • 2000-07-26
    • Joseph Gerard AguilarJuin-Hwey ChenWei WangRobert W. Zopf
    • Joseph Gerard AguilarJuin-Hwey ChenWei WangRobert W. Zopf
    • G10L15/20
    • G10L19/265G10L19/093G10L21/0272G10L25/18G10L25/30G10L25/90G10L25/93
    • 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.
    • 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。