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1. 发明申请

US20090194152A1 THIN-FILM SOLAR CELL HAVING HETERO-JUNCTION OF SEMICONDUCTOR AND METHOD FOR FABRICATING THE SAME 审中-公开
标题翻译：具有半导体异质结的薄膜太阳能电池及其制造方法
公开(公告)号：US20090194152A1
公开(公告)日：2009-08-06
申请号：US12025226
申请日：2008-02-04
申请人： Chee-Wee Liu , Cheng-Yeh Yu , Wen-Yuan Chen , Chu-Hsuan Lin
发明人： Chee-Wee Liu , Cheng-Yeh Yu , Wen-Yuan Chen , Chu-Hsuan Lin
IPC分类号： H01L31/0336 , H01L31/18
CPC分类号： H01L31/022425 , B82Y20/00 , H01L31/022433 , H01L31/028 , H01L31/035254 , H01L31/035281 , H01L31/072 , H01L31/0745 , H01L31/075 , H01L31/1804 , H01L31/1812 , Y02E10/547 , Y02E10/548 , Y02P70/521
摘要： A thin-film solar cell having a hetero-junction of semiconductor and the fabrication method thereof are provided. Instead of the conventional hetero-junction of III-V semiconductor or homo-structure of IV semiconductor, the thin-film solar cell according to the present invention adopts a novel hetero-junction structure of IV semiconductor to improve the cell efficiency thereof. By adjusting the amount of layer sequences and the thickness of the hetero-junction structure, the cell efficiency of the thin-film solar cell according to the present invention is also optimized.
摘要翻译：提供了具有半导体异质结的薄膜太阳能电池及其制造方法。代替III-V族半导体的常规异质结或IV半导体的同构结构，根据本发明的薄膜太阳能电池采用IV半导体的新型异质结结构来提高其电池效率。通过调整层序列的量和异质结结构的厚度，也优化了本发明的薄膜太阳能电池的电池效率。

2. 发明授权

US06792402B1 Method and device for defining table of bit allocation in processing audio signals 失效
标题翻译：用于定义处理音频信号中位分配表的方法和装置
公开(公告)号：US06792402B1
公开(公告)日：2004-09-14
申请号：US09491663
申请日：2000-01-27
申请人： Wen-Yuan Chen
发明人： Wen-Yuan Chen
IPC分类号： G10L1100
CPC分类号： G10L19/035
摘要： A method and a device for defining bit allocation table in processing audio signals are provided. The provided method and device can save storage bits and provide light quality as well. In the first step, the total number of bits for storing audio signals is determined. Then the psychoacoustic model provides many signal-to-mask ratios according to the audio signals. At last, the quantizer quantizes the signal-to-mask ratios to generate several quantized levels each of which corresponds to a bit allocation value to define the table of bit allocation. Therefore, fewer or no storage bits are provided for unimportant subbands and signal frames, that is, the efficiency and quality of transmission of audio signals can be raised.
摘要翻译：提供了一种用于在处理音频信号中定义比特分配表的方法和装置。所提供的方法和设备可以节省存储位并提供光质量。在第一步骤中，确定用于存储音频信号的总位数。然后，心理声学模型根据音频信号提供许多信号与掩模比。最后，量化器对信号与掩模比率进行量化，以产生几个量化电平，每个量化电平对应于位分配值以定义位分配表。因此，为不重要的子带和信号帧提供较少或不存储存储位，即，可以提高音频信号的传输效率和质量。

3. 发明授权

US5970154A Apparatus and method for echo cancellation 失效
标题翻译：回波消除的装置和方法
公开(公告)号：US5970154A
公开(公告)日：1999-10-19
申请号：US876481
申请日：1997-06-16
申请人： Wen-Yuan Chen , Chih-Hung Kuo
发明人： Wen-Yuan Chen , Chih-Hung Kuo
IPC分类号： H04B3/20
CPC分类号： H04M9/082
摘要： Apparatus and method for echo cancellation that incorporate a psychoacoustic model in determining the necessity to update compensator coefficients for the generation of artificial echoes for echo cancellation. The method includes the steps of dividing an audio signal having at least one echo into a plurality of subbands. For each of the plurality of subbands, the echo is reduced based on a variable coefficient transfer function to produce a compensated subband. The signal level of the compensated subband is then compared to a threshold value on a psychoacoustic model curve at a corresponding subband frequency. If the signal level of the compensated subband is greater than the threshold value, the coefficients of the transfer function are updated. Conversely, if the signal level of the compensated subband is less than or equal to the threshold value, the coefficients of the transfer function are not updated.
摘要翻译：用于回声消除的装置和方法，其包括心理声学模型以确定更新补偿器系数以产生用于回波消除的人造回波的必要性。该方法包括将具有至少一个回波的音频信号划分成多个子带的步骤。对于多个子带中的每一个，基于可变系数传递函数来减小回波以产生补偿子带。然后将补偿子带的信号电平与心理声学模型曲线上的相应子带频率处的阈值进行比较。如果补偿子带的信号电平大于阈值，则更新传递函数的系数。相反，如果补偿子带的信号电平小于或等于阈值，则不更新传递函数的系数。

4. 发明授权

US08859380B2 Integrated circuits and manufacturing methods thereof 有权
标题翻译：集成电路及其制造方法
公开(公告)号：US08859380B2
公开(公告)日：2014-10-14
申请号：US12944104
申请日：2010-11-11
申请人： Zhiqiang Wu , Yi-Ming Sheu , Tsung-Hsing Yu , Kuan-Lun Cheng , Chih-Pin Tsao , Wen-Yuan Chen , Chun-Fu Cheng , Chih-Ching Wang
发明人： Zhiqiang Wu , Yi-Ming Sheu , Tsung-Hsing Yu , Kuan-Lun Cheng , Chih-Pin Tsao , Wen-Yuan Chen , Chun-Fu Cheng , Chih-Ching Wang
IPC分类号： H01L29/76 , H01L21/8234
CPC分类号： H01L21/26586 , H01L21/823412 , H01L21/823418 , H01L21/823425 , H01L21/823468 , H01L29/66492 , H01L29/66545 , H01L29/66575
摘要： A method of forming an integrated circuit includes forming a plurality of gate structures longitudinally arranged along a first direction over a substrate. A plurality of angle ion implantations are performed to the substrate. Each of the angle ion implantations has a respective implantation angle with respect to a second direction. The second direction is substantially parallel with a surface of the substrate and substantially orthogonal to the first direction. Each of the implantation angles is substantially larger than 0°.
摘要翻译：形成集成电路的方法包括：在衬底上形成沿着第一方向纵向布置的多个栅极结构。对基板执行多个角度离子注入。每个角度离子注入相对于第二方向具有相应的注入角度。第二方向基本上平行于衬底的表面并且基本上与第一方向正交。每个注入角度基本上大于0°。

5. 发明授权

US06314395B1 Voice detection apparatus and method 失效
标题翻译：语音检测装置及方法
公开(公告)号：US06314395B1
公开(公告)日：2001-11-06
申请号：US09172416
申请日：1998-10-14
申请人： Wen-Yuan Chen
发明人： Wen-Yuan Chen
IPC分类号： G10L1520
CPC分类号： G10L25/87
摘要： A voice detection method and apparatus is provided, which can detect whether a received signal is a voice signal or a background noise. By the method and apparatus, the voice detection need not to perform multiplications and divisions. Moreover, the voice detection method and apparatus can encode the sampled data into 8-bit format but nonetheless obtain good detection result. Further, the voice detection method and apparatus can prevent overflow and allow for easy refreshing of the preset threshold of background noise. These benefits allow the hardware circuitry that implements the voice detection method and apparatus to be significantly simplified in complexity, and thus significantly reduced in manufacturing cost.
摘要翻译：提供一种可以检测接收到的信号是语音信号还是背景噪声的语音检测方法和装置。通过该方法和装置，语音检测不需要执行乘法和除法。此外，语音检测方法和装置可以将采样数据编码为8位格式，但仍能获得良好的检测结果。此外，语音检测方法和装置可以防止溢出并且允许容易地刷新背景噪声的预设阈值。这些优点使得实现语音检测方法和装置的硬件电路在复杂性方面被显着简化，并因此显着地降低了制造成本。

6. 发明授权

US6163614A Pitch shift apparatus and method 失效
标题翻译：换位装置和方法
公开(公告)号：US6163614A
公开(公告)日：2000-12-19
申请号：US972587
申请日：1997-11-18
申请人： Wen-Yuan Chen
发明人： Wen-Yuan Chen
IPC分类号： G10H1/20 , G10H7/02 , H03G5/00
CPC分类号： G10H7/02 , G10H1/20 , G10H2250/631
摘要： A pitch shift apparatus is provided to pitch shift a digital audio signal into a pitch-shifted signal. The apparatus comprises a receiving means, a pitch shifting means and a connecting means, wherein the connecting means comprises: a search region comparator for comparing each sample in the search region with a reference level to obtain a search region bit sequence representing the amplitude of each sample in the search region; a cross region comparator for comparing each sample in the cross region with the reference level to obtain a cross region bit sequence representing the amplitude of each sample in the cross region; a bit processor for bit comparing the cross region bit sequence and any sub-search region bit sequence of M samples in the search region to obtain a corresponding non-similarity; and a connecting device connecting the cross region and a sub-search region corresponding to the minimum non-similarity to renew the pitch-shifted signal.
摘要翻译：音高移位装置用于将数字音频信号俯仰移位成音调移位信号。该装置包括接收装置，音调移位装置和连接装置，其中连接装置包括：搜索区域比较器，用于将搜索区域中的每个采样与参考电平进行比较，以获得表示每个的幅度的搜索区域位序列搜索区域中的样本; 交叉区域比较器，用于将交叉区域中的每个采样与参考电平进行比较，以获得表示交叉区域中每个采样幅度的交叉区域位序列; 比特处理器，用于对搜索区域中的M个样本的交叉区域比特序列和任何子搜索区域比特序列进行比较，以获得相应的非相似性; 以及连接装置，连接所述交叉区域和对应于所述最小非相似性的子搜索区域，以更新所述音调移位信号。

7. 发明授权

US5794191A Neural network based speech recognition method utilizing spectrum-dependent and time-dependent coefficients 失效
标题翻译：基于神经网络的语音识别方法利用频谱依赖和时间依赖系数
公开(公告)号：US5794191A
公开(公告)日：1998-08-11
申请号：US685331
申请日：1996-07-23
申请人： Wen-Yuan Chen
发明人： Wen-Yuan Chen
IPC分类号： G10L15/16 , G10L9/00
CPC分类号： G10L15/16
摘要： An improved artificial neural network for use in speech recognition is disclosed. It comprises an input layer, a hidden layer, and an output layer, each of these layers consisting of a plurality of nodal points. A set of first weighting coefficients are used between the input layer and the hidden layer which are functions of at least one of the nodal points in the hidden layer and at least one of the nodal points in the input layer; whereas, a set of second weighting coefficients, which are functions of time and at least one of the nodal points in the output, are used to correlate between the hidden layer and output layer. In a preferred embodiment, the first weighting coefficients are calculated using the following formula: ##EQU1## i is the index for nodal point in the input layer and a.sub.j, b.sub.j, and c.sub.j are all training coefficients associated with nodal pointj in the hidden layer; and the second weighting coefficients are calculated using the following formula: ##EQU2## n is the timeframe number, r is the order of an orthogonal polynomial series (.psi., 60 .sub.jkm is the m-th order training coefficient between nodal points j and k, in the hidden and output layers, respectively. The use of the two different sets of weighting coefficients allows a timeframe-based division of the speech signals, resulting in a substantial reduction of parameters required for accurate speech recognition.
摘要翻译：公开了一种用于语音识别的改进的人造神经网络。它包括输入层，隐藏层和输出层，这些层中的每一层由多个节点组成。在输入层和隐藏层之间使用一组第一加权系数，其为隐层中的节点中的至少一个和输入层中的至少一个节点的函数; 而作为时间的函数的一组第二加权系数和输出中的节点中的至少一个被用于隐藏层和输出层之间的相关。在优选实施例中，使用以下公式计算第一加权系数： i是输入层中节点的索引，并且aj，bj和cj都是与隐层中的节点j相关联的训练系数; 并且使用以下公式计算第二加权系数： n是时间帧数，r是正交多项式序列的顺序（psi，60jkm是节点j和k之间的第m阶训练系数，在隐藏和输出层中，使用两组不同的加权系数允许语音信号的基于时间帧的划分，导致精确语音识别所需的参数的显着减少。

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