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1. 发明授权

US06182101B2 Method for sampling frequency conversion employing weighted averages 失效
标题翻译：使用加权平均值对变频进行采样的方法
公开(公告)号：US06182101B2
公开(公告)日：2001-01-30
申请号：US08524103
申请日：1995-08-21
申请人： Antti Vahatalo , Olli Ali-Yrkko
发明人： Antti Vahatalo , Olli Ali-Yrkko
IPC分类号： G06F1717
CPC分类号： H03H17/0685
摘要： An improved method converts a first sampling rate of signal values to a second sampling rate, the second sampling rate being a non-integer multiple of the first sampling rate. The method includes the steps of: interpolating between successive signal values occurring at the first sampling rate to derive further signal values at an integer multiple sampling rate thereof, the integer multiple sampling rate falling within a range that is a greater than 0.5 and less than two times an integer multiple of the second sampling rate; employing a method of weighted average interpolation or decimation to derive from the signal values occurring at an integer multiple sampling rate of the first sampling rate, signal value magnitudes occurring at the integer multiple of the second signal sampling rate; and decimating the signal values at the integer multiple of the second sampling rate by an integer multiple factor to derive second sampling rate.
摘要翻译：改进的方法将信号值的第一采样率转换为第二采样率，第二采样率是第一采样率的非整数倍。该方法包括以下步骤：在以第一采样率发生的连续信号值之间进行插值，以得到其整数倍采样率的其它信号值，整数倍采样率落在大于0.5且小于2的范围内乘以第二采样率的整数倍; 采用加权平均插值或抽取方法从以第一采样率的整数倍采样率出现的信号值得到的信号值幅度以第二信号采样率的整数倍发生; 并且以第二采样率的整数倍的信号值抽取整数倍因子以得到第二采样率。

2. 发明授权

US5596677A Methods and apparatus for coding a speech signal using variable order filtering 失效
标题翻译：使用可变顺序滤波对语音信号进行编码的方法和装置
公开(公告)号：US5596677A
公开(公告)日：1997-01-21
申请号：US155574
申请日：1993-11-19
申请人： Kari Jarvinen , Olli Ali-Yrkko
发明人： Kari Jarvinen , Olli Ali-Yrkko
IPC分类号： G10L19/06 , G10L19/12 , G10L3/02
CPC分类号： G10L19/005 , G10L19/06 , G10L19/12 , G10L19/24
摘要： The method concerns digital coding of a speech signal. The method is based on the use of a model of speech production comprising an excitation and shaping of the excitation in a filtering operation in such a manner that the order of the filtering which models the shaping of the excitation signal occurring in the vocal tract is adapted according to the speech signal to be coded. By means of the method it is possible to achieve a total modelling for the speech signal--and thus efficient speech coding--which is better than methods using fixed-order, model-based filtering of the speech tract. From the standpoint of the efficiency of the coding, by decreasing a needlessly large order of the filtering method, the bit rate to be used for coding the excitation signal can be increased or the bit rate resources thus freed up can be allocated for use in the error correction coding. On the other hand, the order of the filtering operation modelling the vocal tract can if necessary be increased if this is of essential benefit in the coding, and correspondingly, the bit rate to be used in coding the excitation signal can be lowered.
摘要翻译：该方法涉及语音信号的数字编码。该方法基于使用语音产生模型，其包括在滤波操作中激励和整形激励，使得对在声道中发生的激励信号的形状进行建模的滤波顺序被适配根据要编码的语音信号。通过该方法，可以实现语音信号的总体建模，从而实现有效的语音编码，这比使用固定次序，基于模型的语音路径滤波方法更好。从编码效率的观点出发，通过减少滤波方法的不必要大的次序，可以增加用于对激励信号进行编码的比特率，或者可以将释放的比特率资源用于纠错编码。另一方面，如果在编码中这是至关重要的，则可以增加对声道进行建模的滤波操作的顺序，并且相应地可以降低用于编码激励信号的比特率。

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