SH2 domain binding inhibitors
    61.
    发明授权
    SH2 domain binding inhibitors 失效
    SH2结构域结合抑制剂

    公开(公告)号:US06977241B2

    公开(公告)日:2005-12-20

    申请号:US10362231

    申请日:2001-08-22

    摘要: Disclosed are compounds for SH2 domain binding inhibition. For example, disclosed is a compound of formula (I) wherein R1 is a lipophile; R2, in combination with the phenyl ring, forms a phenylphosphate mimic group or a protected phenylphosphate mimic group; R3 is hydrogen, azido, amino, carboxyalkyl, alkoxycarbonylalkyl, aminocarbonylalkyl, or alkylcarbonylamino, wherein the alkyl portion of R3 may be optionally substituted with a substituent selected from the group consisting of halo, hydroxy, carboxyl, amino, aminoalkyl, alkyl, alkoxy, and keto; R6 is a linker; AA is an amino acid; and n is 1 to 6; or a salt thereof. Also disclosed are a pharmaceutical composition, a method for inhibiting an SH2 domain from binding with a phosphoprotein and a method of treating breast cancer.

    摘要翻译: 公开了用于SH2结构域结合抑制的化合物。 例如,公开了式(I)的化合物,其中R 1是亲脂体; R 2与苯环组合形成苯基磷酸酯模拟基团或被保护的苯基磷酸酯模拟基团; R 3是氢,叠氮基,氨基,羧基烷基,烷氧基羰基烷基,氨基羰基烷基或烷基羰基氨基,其中R 3的烷基部分可以任选地被选自 由卤素,羟基,羧基,氨基,氨基烷基,烷基,烷氧基和酮基组成; R 6是连接体; AA是氨基酸; n为1〜6; 或其盐。 还公开了药物组合物,抑制SH2结构域与磷蛋白结合的方法和治疗乳腺癌的方法。

    Coding based on spectral content of a speech signal
    62.
    发明授权
    Coding based on spectral content of a speech signal 有权
    基于语音信号的频谱内容进行编码

    公开(公告)号:US06937979B2

    公开(公告)日:2005-08-30

    申请号:US09896682

    申请日:2001-06-29

    申请人: Yang Gao Huan-Yu Su

    发明人: Yang Gao Huan-Yu Su

    IPC分类号: G10L19/14 G10L21/02 G10L19/00

    摘要: In a coding procedure, a spectral content of a speech signal is estimated. A preferential coding algorithm or preferential value of at least one coding parameter is selected based on the estimated spectral content of the speech signal. The speech signal is coded in accordance with the selected coding algorithm or the selected coding parameter to control the operation of one or more of the following: a pre-processing filter, a post-processing filter, a coding control coefficient, a weighting filter, a synthesis filter, and a quantization table.

    摘要翻译: 在编码过程中,估计语音信号的频谱内容。 基于所估计的语音信号的频谱内容来选择优选编码算法或至少一个编码参数的优先值。 语音信号根据所选择的编码算法或选择的编码参数进行编码,以控制以下一个或多个的操作:预处理滤波器,后处理滤波器,编码控制系数,加权滤波器, 合成滤波器和量化表。

    Simple noise suppression model
    63.
    发明申请
    Simple noise suppression model 有权
    简单的噪声抑制模型

    公开(公告)号:US20050065792A1

    公开(公告)日:2005-03-24

    申请号:US10799505

    申请日:2004-03-11

    申请人: Yang Gao

    发明人: Yang Gao

    摘要: An approach for efficiently reducing background noise from speech signal in real-time applications is presented. A noisy input speech signal is processed through an inverse filter when the spectrum tilt of the input signal is not that of a pure background noise model the noisy input signal is also filtered in order to reduce the spectrum valley areas of the noisy input signal when the background noise is present.

    摘要翻译: 提出了一种在实时应用中有效降低语音信号背景噪声的方法。 当输入信号的频谱倾斜不是纯背景噪声模型的频谱倾斜时,噪声输入语音信号通过反向滤波器被处理,噪声输入信号也被滤波,以便当噪声输入信号的频谱谷谷区域减小时,噪声输入信号 背景噪音存在。

    Codebook tables for multi-rate encoding and decoding with pre-gain and delayed-gain quantization tables
    64.
    发明授权
    Codebook tables for multi-rate encoding and decoding with pre-gain and delayed-gain quantization tables 有权
    用于具有预增益和延迟增益量化表的多速率编码和解码的码表

    公开(公告)号:US06757649B1

    公开(公告)日:2004-06-29

    申请号:US10409404

    申请日:2003-04-08

    IPC分类号: G10L1912

    摘要: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

    摘要翻译: 公开了能够将语音信号编码为比特流以进行后续解码以产生合成语音的语音压缩系统。 语音压缩系统通过将期望的平均比特率与重构语音的感知质量进行平衡来优化比特流消耗的带宽。 语音压缩系统包括全速率编解码器,半速率编解码器,四分之一速率编解码器和八速率编解码器。 基于速率选择来选择性地激活编解码器。 此外,基于类型分类,全速率和半速率编解码器被选择性地激活。 选择性地激活每个编解码器以以强调语音信号的不同方面的不同比特率对语音信号进行编码和解码,以增强合成语音的整体质量。

    Pitch determination using speech classification and prior pitch estimation
    65.
    发明授权
    Pitch determination using speech classification and prior pitch estimation 有权
    使用语音分类和先前音调估计的音调确定

    公开(公告)号:US06507814B1

    公开(公告)日:2003-01-14

    申请号:US09154654

    申请日:1998-09-18

    申请人: Yang Gao

    发明人: Yang Gao

    IPC分类号: G10L1904

    摘要: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To achieve high quality in lower bit rate encoding modes, the speech encoder departs from the strict waveform matching criteria of regular CELP coders and strives to identify significant perceptual features of the input signal. To support lower bit rate encoding modes, a variety of techniques are applied many of which involve the classification of the input signal. For each bit rate mode selected, pluralities of fixed or innovation subcodebooks are selected for use in generating innovation vectors. The speech encoder also utilizes an adaptive weighting factor in the selection of a current pitch lag value from a plurality of pitch lag candidates. For example, if the speech encoder identifies an integer multiple timing relationship between any two pitch lag candidates, the pitch lag candidate with the smallest timing value is favored through adjustment of the weighting factor. Similarly, if a pitch lag candidate exhibits timing that corresponds to that of previous pitch lag values, the weighting factor is adjusted to favor that candidate.

    摘要翻译: 多速率语音编解码器通过自适应地选择编码比特率模式以匹配通信信道限制来支持多种编码比特率模式。 在较高的比特率编码模式中,通过CELP(码激励线性预测)和其他相关联的建模参数的语音的精确表示被生成用于更高质量的解码和再现。 为了在低比特率编码模式下实现高质量,语音编码器脱离了常规CELP编码器的严格波形匹配标准,并努力识别输入信号的重要感知特征。 为了支持较低比特率编码模式,应用了许多技术,其中许多技术涉及输入信号的分类。 对于所选择的每个比特率模式,选择多个固定或创新子码本来用于产生创新向量。 语音编码器还在从多个音调滞后候选中选择当前音调滞后值的同时利用自适应加权因子。 例如,如果语音编码器识别任何两个音调滞后候选之间的整数倍定时关系,则通过调整加权因子,有利于具有最小定时值的音调滞后候选。 类似地,如果音调滞后候选呈现对应于先前音调滞后值的定时,则调整加权因子以有利于该候选。

    Speech encoder using gain normalization that combines open and closed loop gains
    66.
    发明授权
    Speech encoder using gain normalization that combines open and closed loop gains 有权
    使用组合开环和闭环增益的增益归一化的语音编码器

    公开(公告)号:US06260010B1

    公开(公告)日:2001-07-10

    申请号:US09156650

    申请日:1998-09-18

    IPC分类号: G10L1900

    摘要: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To support lower bit rate encoding modes, a variety of techniques are applied many of which involve the classification of the input signal. The encoder utilizes gain normalization wherein LPC (linear predictive coding) gain provides a smoothing factor for combining both open and closed loop gains. The lower the LPC gain, the greater the open loop gain contribution to a gain normalization factor. The greater the LPC gain, the greater the closed loop gain contribution. For background noise, the smaller of the closed and open loop gains are used as the normalization factor. The normalization factor is limited by the LPC gain to prevent influencing the coding quality.

    摘要翻译: 多速率语音编解码器通过自适应地选择编码比特率模式以匹配通信信道限制来支持多种编码比特率模式。 在较高的比特率编码模式中,通过CELP(码激励线性预测)和其他相关联的建模参数的语音的精确表示被生成用于更高质量的解码和再现。 为了支持较低比特率编码模式,应用了许多技术,其中许多技术涉及输入信号的分类。 编码器利用增益归一化,其中LPC(线性预测编码)增益提供用于组合开路和闭环增益的平滑因子。 LPC增益越低,开环增益对增益归一化因子的贡献越大。 LPC增益越大,闭环增益贡献越大。 对于背景噪声,使用较小的闭环和开环增益作为归一化因子。 归一化因子受LPC增益的限制,以防止影响编码质量。

    Energy adjustment of acoustic echo replica signal for speech enhancement
    67.
    发明授权
    Energy adjustment of acoustic echo replica signal for speech enhancement 有权
    用于语音增强的声学回波复制信号的能量调节

    公开(公告)号:US09589556B2

    公开(公告)日:2017-03-07

    申请号:US14740686

    申请日:2015-06-16

    申请人: Yang Gao

    发明人: Yang Gao

    摘要: A method for cancelling/reducing acoustic echoes in speech/audio signal enhancement processing comprises using a received reference signal to excite an adaptive filter wherein the output of the adaptive filter forms a replica signal of acoustic echo; the replica signal of acoustic echo is reduced adaptively by multiplying a gain to get a gained replica signal of acoustic echo wherein the gain is smaller in speech area and/or double-talk area than non-speech area; the gained replica signal of acoustic echo is subtracted from a microphone input signal to suppress the acoustic echo in the microphone input signal.

    摘要翻译: 用于消除/减少语音/音频信号增强处理中的声学回声的方法包括使用接收的参考信号来激励自适应滤波器,其中自适应滤波器的输出形成声学回声的复制信号; 通过乘以增益自适应地减小声学回波的复制信号,以获得声学回波的增益复制信号,其中语音区域和/或双方话区域中的增益比非语音区域小; 从麦克风输入信号中减去所获得的声学回波的复制信号,以抑制麦克风输入信号中的声学回声。

    Post tone suppression for speech enhancement
    68.
    发明授权
    Post tone suppression for speech enhancement 有权
    用于语音增强的音调抑制

    公开(公告)号:US09520139B2

    公开(公告)日:2016-12-13

    申请号:US14740735

    申请日:2015-06-16

    申请人: Yang Gao

    发明人: Yang Gao

    摘要: A method for reducing disturbing tone signals from acoustic echoes or background noises in speech/audio signal enhancement processing comprises decomposing a full-band input microphone speech signal into plurality of sub-band component channel signals by using a filter-bank; disturbing tone signal in each sub-band component channel signal is detected by using parameters such as a second reflection coefficient from LPC analysis, a time domain sharpness parameter, a normalized pitch correlation, and/or a SNR parameter; the energy of the current sub-band component channel signal is reduced by multiplying a reduction gain if the disturbing tone signal is detected in the current sub-band component channel signal; all the component channel signals are summed back to output a tone-reduced full band speech signal.

    摘要翻译: 用于在语音/音频信号增强处理中减少来自声学回声或背景噪声的干扰音信号的方法包括:通过使用滤波器组将全频带输入麦克风语音信号分解为多个子带分量信道信号; 通过使用诸如来自LPC分析的第二反射系数,时域锐度参数,归一化音调相关性和/或SNR参数的参数来检测每个子带分量信道信号中的干扰音信号。 如果在当前子带分量信道信号中检测到干扰音信号,则通过乘以减小增益来减小当前子带分量信道信号的能量; 所有分量信道信号被相加以输出音调降低的全频带语音信号。

    Method and system for telecommunication network to provide session service to internet
    70.
    发明申请
    Method and system for telecommunication network to provide session service to internet 审中-公开
    电信网络为互联网提供会话服务的方法和系统

    公开(公告)号:US20150334136A1

    公开(公告)日:2015-11-19

    申请号:US14429482

    申请日:2012-08-30

    申请人: Yang Gao Lingjiang Mu

    发明人: Yang Gao Lingjiang Mu

    IPC分类号: H04L29/06 H04L29/12 H04L29/08

    摘要: Provided is a method for a telecommunication network to provide a session service to the Internet. An access gateway of a telecommunication network supports an Internet application protocol, or an access-side device of the telecommunication network is upgraded to support the Internet application protocol, and the Internet is connected to the telecommunication network via the access gateway or the upgraded access-side device. The method further includes that an Internet user establishes a session with a telecommunication user or another Internet user via a telecommunication network. A system for a telecommunication network to provide a session service to the Internet is also provided. By upgrading an access-side device of a telecommunication network, the disclosure enables the access-side device to support an Internet application protocol and converges the Internet with the telecommunication network, thus being able to provide a session service to an Internet user. The disclosure facilitates the usage of the session service by the Internet user.

    摘要翻译: 提供了一种用于电信网络向互联网提供会话服务的方法。 电信网络的接入网关支持互联网应用协议,或者电信网络的接入侧设备被升级以支持因特网应用协议,并且因特网通过接入网关或升级的接入网络连接到电信网络, 侧设备。 该方法还包括因特网用户通过电信网络与电信用户或其他因特网用户建立会话。 还提供了一种用于电信网络向因特网提供会话服务的系统。 通过升级电信网络的接入侧设备,本公开使得接入侧设备能够支持因特网应用协议,并将因特网与电信网络融合,从而能够向因特网用户提供会话服务。 本公开有利于因特网用户使用会话服务。