Gain quantization for a CELP speech coder
    51.
    发明授权
    Gain quantization for a CELP speech coder 有权
    为CELP语音编码器增益量化

    公开(公告)号:US07260522B2

    公开(公告)日:2007-08-21

    申请号:US10888420

    申请日:2004-07-10

    IPC分类号: G10L19/00

    摘要: There is provided a speech encoding system that receives a speech signal. The speech encoding system comprises a frame processor for processing a frame of the speech signal, where the frame processor includes a pitch gain generator that derives unquantized pitch gains, and a first vector guantizer that receives the unquantized pitch gains and generates quantized pitch gains. The speech encoding system also comprises a subframe processor that begins subframe processing after the pitch gain generator has derived the unquantized pitch gains and the first vector quantizer has generated the quantized pitch gains.

    摘要翻译: 提供了一种接收语音信号的语音编码系统。 语音编码系统包括用于处理语音信号的帧的帧处理器,其中帧处理器包括导出非量化音调增益的音调增益发生器,以及接收未量化音调增益并产生量化音调增益的第一矢量指示器。 语音编码系统还包括在音调增益发生器已经导出无量子化音调增益并且第一矢量量化器已经产生量化音调增益之后开始子帧处理的子帧处理器。

    Adaptive voice mode extension for a voice activity detector
    53.
    发明申请
    Adaptive voice mode extension for a voice activity detector 有权
    语音活动检测器的自适应语音模式扩展

    公开(公告)号:US20060217973A1

    公开(公告)日:2006-09-28

    申请号:US11342104

    申请日:2006-01-26

    IPC分类号: G10L19/12

    CPC分类号: G10L25/78 G10L2025/786

    摘要: There is provided a voice activity detection method for indicating an active voice mode and an inactive voice mode. The method comprises receiving a first portion of an input signal; determining that the first portion of the input signal includes an active voice signal; indicating the active voice mode in response to the determining that the first portion of the input signal includes the active voice signal; receiving a second portion of the input signal immediately following the first portion of the input signal; determining that the second portion of the input signal includes an inactive voice signal; extending the indicating the active voice mode for a period of time after determining that the second portion of the input signal includes the inactive voice signal, wherein the period of time varies based on one or more conditions; and indicating the inactive voice mode after expiration of the period of time.

    摘要翻译: 提供了一种用于指示主动语音模式和无效语音模式的语音活动检测方法。 该方法包括接收输入信号的第一部分; 确定输入信号的第一部分包括有效语音信号; 响应于确定输入信号的第一部分包括有效语音信号,指示主动语音模式; 接收紧接在输入信号的第一部分之后的输入信号的第二部分; 确定输入信号的第二部分包括不活动的语音信号; 在确定所述输入信号的第二部分包括所述不活动语音信号之后,将所述主动语音模式指示一段时间,其中所述时间段基于一个或多个条件而变化; 并且在该时间段期满之后指示不活动的语音模式。

    Fixed code book with embedded adaptive code book
    54.
    发明授权
    Fixed code book with embedded adaptive code book 有权
    固定代码本嵌入式自适应代码簿

    公开(公告)号:US07103538B1

    公开(公告)日:2006-09-05

    申请号:US10166315

    申请日:2002-06-10

    申请人: Yang Gao

    发明人: Yang Gao

    IPC分类号: G10L19/14

    CPC分类号: G10L19/08 G10L2019/0007

    摘要: A system including an adaptive code book and a fixed code book for code excited linear prediction coding of speech signals is provided. The invention includes an embedded adaptive code book in the fixed code book and the selection procedure for selecting excitation vector parameters. A code book update system updates the fixed code book with embedded adaptive code book based on the long term processing excitation vector parameters with previous synthesized excitation.

    摘要翻译: 提供一种包括自适应码本和用于语音信号的码激励线性预测编码的固定码本的系统。 本发明包括固定码本中的嵌入式自适应码本和用于选择激励矢量参数的选择过程。 代码簿更新系统基于具有先前合成激励的长期处理激励矢量参数,利用嵌入式自适应码本更新固定码本。

    SH2 domain binding inhibitors
    55.
    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
    56.
    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
    57.
    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
    58.
    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
    59.
    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
    60.
    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。