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    Phrase splicing and variable substitution using a trainable speech synthesizer
    1.
    发明授权
    Phrase splicing and variable substitution using a trainable speech synthesizer 有权
    短语拼接和可变替代使用可训练的语音合成器

    公开(公告)号：US06266637B1

    公开(公告)日：2001-07-24

    申请号：US09152178

    申请日：1998-09-11

    申请人： Robert E. Donovan ,  Martin Franz ,  Salim E. Roukos ,  Jeffrey Sorensen

    发明人： Robert E. Donovan ,  Martin Franz ,  Salim E. Roukos ,  Jeffrey Sorensen

    IPC分类号： G10L1300

    CPC分类号： G10L13/06

    摘要： In accordance with the present invention, a method for providing generation of speech includes the steps of providing input to be acoustically produced, comparing the input to training data or application specific splice files to identify one of words and word sequences corresponding to the input for constructing a phone sequence, using a search algorithm to identify a segment sequence to construct output speech according to the phone sequence and concatenating segments and modifying characteristics of the segments to be substantially equal to requested characteristics. Application specific data is advantageously used to make pertinent information available to synthesize both the phone sequence and the output speech. Also, described is a system for performing operations in accordance with the disclosure.

    摘要翻译： 根据本发明，一种用于提供语音产生的方法包括以下步骤：提供要声学产生的输入，将输入与训练数据或应用特定拼接文件进行比较，以识别与用于构建的输入相对应的单词和单词序列之一 电话序列，使用搜索算法来识别段序列以根据电话序列构建输出语音，并且将段并且将段的特征修改为基本等于请求的特征。 应用特定数据有利地用于使相关信息可用于合成电话序列和输出语音。 此外，描述了根据本公开的用于执行操作的系统。

    Adaptation of statistical parsers based on mathematical transform
    2.
    发明授权
    Adaptation of statistical parsers based on mathematical transform 有权
    基于数学变换的统计解析器的适应

    公开(公告)号：US07308400B2

    公开(公告)日：2007-12-11

    申请号：US09737259

    申请日：2000-12-14

    申请人： Xiaoqiang Luo ,  Salim E. Roukos ,  Robert T. Ward

    发明人： Xiaoqiang Luo ,  Salim E. Roukos ,  Robert T. Ward

    IPC分类号： G06F17/27 ,  G10L15/18

    CPC分类号： G06F17/2715

    摘要： An arrangement for adapting statistical parsers to new data using a mathematical transform, particularly a Markov transform. In particular, it is assumed that an initial statistical parser is available and a batch of new data is given. The initial model is mapped to a new model by a Markov matrix, each of whose rows sums to one. In the unsupervised setup, where “true” parses are missing, the transform matrix is obtained by maximizing the log likelihood of the parses of test data decoded using the model before adaptation. The proposed algorithm can be applied to supervised adaptation, as well.

    摘要翻译： 使用数学变换，特别是马尔科夫变换，使统计解析器适应新数据的安排。 特别地，假设初始统计解析器可用并且给出一批新数据。 初始模型通过马尔可夫矩阵映射到新模型，每个行的行总和为1。 在无人监控的设置中，“真”解析丢失，通过最大化在适应之前使用模型解码的测试数据的解析的对数似然性来获得变换矩阵。 所提出的算法也可以应用于监督适应。

    Embedded signalling
    6.
    发明授权
    Embedded signalling 失效
    嵌入式信号

    公开(公告)号：US5319735A

    公开(公告)日：1994-06-07

    申请号：US808913

    申请日：1991-12-17

    申请人： Robert D. Preuss ,  Salim E. Roukos ,  A. W. F. Huggins ,  Herbert Gish ,  Marcos A. Bergamo ,  Patrick M. Peterson ,  Alan G. Derr

    发明人： Robert D. Preuss ,  Salim E. Roukos ,  A. W. F. Huggins ,  Herbert Gish ,  Marcos A. Bergamo ,  Patrick M. Peterson ,  Alan G. Derr

    IPC分类号： G10L19/00 ,  G11B20/00 ,  G11B20/10 ,  G11B20/18 ,  H03M5/04 ,  H04B1/66 ,  H04B1/707 ,  H04L5/00 ,  H04N5/92 ,  G10L3/00

    CPC分类号： G11B20/00913 ,  G10L19/018 ,  G11B20/00086 ,  G11B20/10009 ,  G11B20/10527 ,  G11B20/1813 ,  H03M5/04 ,  H04B1/665 ,  H04B1/707

    摘要： A code signal representing a sequence of code symbols carrying digital information is generated with the frequency components of the code signal being essentially confined to a preselected signalling band lying within the bandwidth of an audio signal within which the code signal is to be embedded. The audio signal is continuously frequency analyzed over a frequency band encompassing the signalling band and the code signal is dynamically filtered as a function of the analysis thereby to provide a modified code signal with frequency components which are, at each time instant, essentially a preselected small proportion of the levels of the corresponding audio signal frequency components. Accordingly, the modified code signal can be combined with the audio signal to obtain a composite audio signal which is not readily distinguishable from the original audio signal by listening. Furthermore, the digital information can be recovered from the composite audio signal by a procedure which is essentially the compliment of that used to obtain the composite audio signal.

    摘要翻译： 产生代表携带数字信息的码元序列的代码信号，其中代码信号的频率分量基本上被限制在位于要嵌入代码信号的音频信号的带宽内的预选信令频带。 音频信号在包围信令频带的频带上被连续频率分析，并且代码信号作为分析的函数被动态滤波，从而提供具有频率分量的修改的码信号，频率分量基本上是预选小的 相应音频信号频率分量的比例。 因此，可以将修改的代码信号与音频信号组合以获得不容易通过收听与原始音频信号区分的复合音频信号。 此外，可以通过本质上是用于获得复合音频信号的补充的过程从复合音频信号中恢复数字信息。