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    • 1. 发明专利
    • Method and system for testing speech intelligibility in children
    • AU2002243790B2
    • 2005-12-08
    • AU2002243790
    • 2002-02-04
    • WISCONSIN ALUMNI RES FOUND
    • LITOVSKY RUTH Y
    • A61B5/12G10L13/04G09B007/08G10L021/00A61B005/00
    • A method and system for testing the speech intelligibility of a child comprises providing a set of target sounds as words in the presence and absence of competing sound(s) of a variety of types so as to enable an analysis of the aspects of competing sounds and their respective effects on the speech intelligibility of a child. Locations at which competing sound(s) is provided is varied to enable an evaluation of its effect on the spatial release from masking. The target words used in the test are first determined to be within the vocabulary of the child. The child is required to respond to the target word by selecting a picture representation of the target word from among several picture choices, thus providing an interactive aspect to the test. There may optionally be provided a positive or a negative reinforcement. The sound level at which the target words are presented may vary adaptively according to the child's responses, the change in sound level being determined by a set of rules. The test is repeated over several target words and under a variety of types and locations of competing sounds and the child's responses recorded in a results database. The results are available for further analysis by a user to produce customized output. A computerized system is disclosed that enables the provisioning of the test in a controlled manner, analysis of the data and further engagement the child.
    • 3. 发明申请
    • Architecture for a speech input method editor for handheld portable devices
    • 用于手持便携式设备的语音输入法编辑器的架构
    • US20040243415A1
    • 2004-12-02
    • US10452429
    • 2003-06-02
    • International Business Machines Corporation
    • Patrick M. CommarfordMario E. De ArmasBurn L. LewisJames R. Lewis
    • G10L021/00
    • G06F3/167G10L15/26
    • A speech input method editor can include a speech toolbar (102) having at least a microphone state/toggle button (104). The speech input method editor can also include a selectable dictation window area (108) used as a temporary dictation target until dictation text is transferred to a target application and a selectable correction window area (112) having at least one among an alternate list (120) for correcting dictated words, an alphabet (114), a spacebar (116), a spell mode reminder (118), or a virtual keyboard (122). The speech input method editor can remain active while using the selectable correction window and while transferring dictation text to the target application. The speech input method editor can further include an alternate input method editor window (112b) used to allow non-speech editing into at least one among the dictation window or to the target application while using the speech input method editor.
    • 语音输入法编辑器可以包括具有至少麦克风状态/切换按钮(104)的语音工具栏(102)。 语音输入方法编辑器还可以包括用作临时听写目标的可选听写窗口区域(108),直到听写文本被传送到目标应用程序,并且可选修正窗口区域(112)具有备选列表(120 ),用于校正指定词,字母表(114),空格键(116),拼写模式提醒(118)或虚拟键盘(122)。 语音输入法编辑器可以在使用可选择的校正窗口和将录音文本传送到目标应用程序时保持有效。 语音输入法编辑器还可以包括用于在使用语音输入方法编辑器时允许在听写窗口或目标应用程序中的至少一个语音编辑的替代输入法编辑器窗口(112b)。
    • 5. 发明申请
    • Method and device for processing time-discrete audio sampled values
    • 用于处理时间离散音频采样值的方法和设备
    • US20040220805A1
    • 2004-11-04
    • US10479398
    • 2004-06-25
    • Ralf GeigerThomas SporerJurgen KollerKarlheinz BrandenburgJurgen Herre
    • G10L019/14G10L021/00
    • G10L19/0212G06F17/147
    • In order to obtain an integer transform, which provides integer output values, the TDAC function of a MDCT is explicitly carried out in the time domain before the forward transform. In overlapping windows, this results in a Givens rotation which may be represented by lifting matrices, wherein time-discrete sampled values of an audio signal may at first be summed up on a pair-wise basis to build a vector so as to be sequentially provided with a lifting matrix. In accordance with the invention, after each multiplication of a vector by a lifting matrix, a rounding step is carried out such that, on the output-side, only integers will result. By transforming the windowed integer sampled value with an integer transform, a spectral representation with integer spectral values may be obtained. The inverse mapping with an inverse rotation matrix and corresponding inverse lifting matrices results in an exact reconstruction. The inventive concept provides a lossless transform which may be coupled immediately with an entropy-encoder without quantizing so as to obtain a windowing and transform method which may be favorably implemented on a hardware-basis.
    • 为了获得提供整数输出值的整数变换,MDCT的TDAC功能在正向变换之前的时域中被明确地执行。 在重叠窗口中,这导致Givens旋转,其可以由提升矩阵表示,其中音频信号的时间离散采样值可以首先在成对的基础上相加以构建向量以便顺序地提供 与提升矩阵。 根据本发明,在通过提升矩阵对向量进行每次乘法之后,执行舍入步骤,使得在输出侧仅将导致整数。 通过用整数变换变换窗口整数采样值,可以获得具有整数频谱值的频谱表示。 具有逆旋转矩阵和对应的反提升矩阵的逆映射导致精确重建。 本发明的概念提供了一种无损变换,其可以与熵编码器立即耦合而不进行量化,以获得可以在硬件上有利地实现的加窗和变换方法。
    • 8. 发明申请
    • Speech-to-speech generation system and method
    • 语音到语音生成系统和方法
    • US20040172257A1
    • 2004-09-02
    • US10683335
    • 2003-10-10
    • International Business Machines Corporation
    • Shen LiqinShi QinDonald T. TangZhang Wei
    • G10L021/00
    • G10L13/00G10L13/04
    • An expressive speech-to-speech generation system and method which can generate expressive speech output by using expressive parameters extracted from the original speech signal to drive the standard TTS system. The system comprises: speech recognition means, machine translation means, text-to-speech generation means, expressive parameter detection means for extracting expressive parameters from the speech of language A, and expressive parameter mapping means for mapping the expressive parameters extracted by the expressive parameter detection means from language A to language B, and driving the text-to-speech generation means by the mapping results to synthesize expressive speech. The system and method can improve the quality of the speech output of the translating system or TTS system.
    • 一种具有表现力的语音对语音生成系统和方法,可以通过使用从原始语音信号中提取的表现力参数来产生表达性语音输出来驱动标准TTS系统。 该系统包括:语音识别装置,机器翻译装置,文本到语音生成装置,用于从语言A的语音中提取表达性参数的表现参数检测装置,以及用于映射表达参数提取的表现参数的表现参数映射装置 从语言A到语言B的检测手段,并且通过映射结果来驱动文本到语音生成装置以合成表达性语音。 系统和方法可以提高翻译系统或TTS系统语音输出的质量。
    • 10. 发明申请
    • Language learning system and a digital storage unit
    • 语言学习系统和数字存储单元
    • US20040148175A1
    • 2004-07-29
    • US10479046
    • 2004-02-23
    • Eero NieminenTimo Saarni
    • G10L021/00
    • G09B17/00G09B19/04
    • Audio recordings and programs are saved as audio files on a digital storage unit in a language learning system comprising student units connected to the digital storage unit. The digital storage unit is provided with a audio interface controller (201) having a dedicated input/output RAM buffer (B1-B63) for each student station. Each RAM buffer has an associated file which is either a fixed file or can be defined for each case. When the audio interface controller (201) receives a record command relating to a specific buffer, the controller (201) opens an audio file associated with the specific buffer, buffers the audio data received from a student station or another source in the buffer, and transfers the contents of the specified buffer to the opened associated audio file. Similarly, in response to a play command relating to a specific RAM buffer, the controller (201) opens an associated audio file in the digital storage unit, transfers audio data from the opened audio file to the buffer, and sends the audio data from the buffer to a respective student station or to other destination.