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1. 发明申请

US20080275695A1 Method and system for pitch contour quantization in audio coding 有权
标题翻译：音频编码中音调轮廓量化的方法和系统
公开(公告)号：US20080275695A1
公开(公告)日：2008-11-06
申请号：US12150307
申请日：2008-04-25
申请人： Anssi Ramo , Jani Nurminen , Sakari Himanen , Ari Heikkinen
发明人： Anssi Ramo , Jani Nurminen , Sakari Himanen , Ari Heikkinen
IPC分类号： G10L11/04
CPC分类号： G10L19/032 , G10L19/09
摘要： A method and device for improving coding efficiency in audio coding. From the pitch values of a pitch contour of an audio signal, a plurality of simplified pitch contour segments are generated to approximate the pitch contour, based on one or more pre-selected criteria. The contour segments can be linear or non-linear with each contour segment represented by a first end point and a second end point. If the contour segments are linear, then only the information regarding the end points, instead of the pitch values, are provided to a decoder for reconstructing the audio signal. The contour segment can have a fixed maximum length or a variable length, but the deviation between a contour segment and the pitch values in that segment is limited by a maximum value.
摘要翻译：一种提高音频编码效率的方法和装置。根据音频信号的音调轮廓的音调值，基于一个或多个预先选择的标准，生成多个简化俯仰轮廓线段以近似俯仰轮廓。轮廓段可以是由第一终点和第二终点表示的每个轮廓段线性或非线性的。如果轮廓段是线性的，则仅将关于终点而不是音调值的信息提供给用于重建音频信号的解码器。轮廓段可以具有固定的最大长度或可变长度，但轮廓段与该段中的俯仰值之间的偏差受到最大值的限制。

2. 发明申请

US20080235009A1 Method and apparatus for reducing synchronization delay in packet switched voice terminals using speech decoder modification 有权
标题翻译：使用语音解码器修改来减少分组交换语音终端中的同步延迟的方法和装置
公开(公告)号：US20080235009A1
公开(公告)日：2008-09-25
申请号：US12154487
申请日：2008-05-23
申请人： Ari Heikkinen , Ari Lakaniemi
发明人： Ari Heikkinen , Ari Lakaniemi
IPC分类号： G10L19/00
CPC分类号： G10L19/167 , G10L21/04 , G10L2019/0012 , H04J3/0632
摘要： A device is disclosed that makes packetized and encoded speech data audible to a listener, as is a method for operating the device. The device includes a unit for generating a synchronization request for reducing an amount of synchronization delay, and further includes a speech decoder that is responsive to the synchronization delay adjustment request for executing a time-warping operation for one of lengthening or shortening a duration of a speech frame. In one embodiment the speech decoder comprises a code excited linear prediction (CELP) speech decoder, and the CELP decoder time-warping operation is applied to a reconstructed excitation signal u(k) to derive a time-warped reconstructed signal uw(k). The time-warped reconstructed signal uw(k) is input to a Linear Predictor (LP) synthesis filter to derive a CELP decoder time-warped output signal ŷw(k) In another embodiment the speech decoder comprises a parametric speech decoder, and where an adaptation of the frame length N in the parametric speech decoder results in the use of a modified frame length Nw.
摘要翻译：公开了一种使分组化和编码的语音数据可听见的收听器的设备，以及用于操作设备的方法。该装置包括用于产生用于减少同步延迟量的同步请求的单元，并且还包括语音解码器，该语音解码器响应于同步延迟调整请求，用于执行延时或缩短持续时间的延时操作语音框架在一个实施例中，语音解码器包括代码激励线性预测（CELP）语音解码器，并且将CELP解码器时间扭曲操作应用于重构的激励信号u（k）以导出时间扭曲的重构信号u （k）。时间扭曲重构信号u（k）被输入到线性预测器（LP）合成滤波器，以导出CELP解码器时间扭曲输出信号ŷŷŷ ）。在另一个实施例中，语音解码器包括参数语音解码器，并且其中参数语音解码器中的帧长度N的调整导致使用经修改的帧长度N N w N。

3. 发明授权

US07394833B2 Method and apparatus for reducing synchronization delay in packet switched voice terminals using speech decoder modification 有权
标题翻译：使用语音解码器修改来减少分组交换语音终端中的同步延迟的方法和装置
公开(公告)号：US07394833B2
公开(公告)日：2008-07-01
申请号：US10364588
申请日：2003-02-11
申请人： Ari Heikkinen , Ari Lakaniemi
发明人： Ari Heikkinen , Ari Lakaniemi
IPC分类号： H04J3/06
CPC分类号： G10L19/167 , G10L21/04 , G10L2019/0012 , H04J3/0632
摘要： A device is disclosed that makes packetized and encoded speech data audible to a listener, as is a method for operating the device. The device includes a unit for generating a synchronization request for reducing an amount of synchronization delay, and further includes a speech decoder that is responsive to the synchronization delay adjustment request for executing a time-warping operation for one of lengthening or shortening a duration of a speech frame. In one embodiment the speech decoder comprises a code excited linear prediction (CELP) speech decoder, and the CELP decoder time-warping operation is applied to a reconstructed excitation signal u(k) to derive a time-warped reconstructed signal uw(k). The time-warped reconstructed signal uw(k) is input to a Linear Predictor (LP) synthesis filter to derive a CELP decoder time-warped output signal y^w(k). In another embodiment the speech decoder comprises a parametric speech decoder, and where an adaptation of the frame length N in the parametric speech decoder results in the use of a modified frame length Nw.
摘要翻译：公开了一种使分组化和编码的语音数据可听见的收听器的设备，以及用于操作设备的方法。该装置包括用于产生用于减少同步延迟量的同步请求的单元，并且还包括语音解码器，该语音解码器响应于同步延迟调整请求，用于执行延时或缩短持续时间的延时操作语音框架在一个实施例中，语音解码器包括代码激励线性预测（CELP）语音解码器，并且将CELP解码器时间扭曲操作应用于重构的激励信号u（k）以导出时间扭曲的重构信号u （k）。时间扭曲重构信号u（k）被输入到线性预测器（LP）合成滤波器，以导出CELP解码器时间扭曲输出信号。（k）。在另一个实施例中，语音解码器包括参数语音解码器，并且其中参数语音解码器中的帧长度N的调整导致使用经修改的帧长度N N w N。

4. 发明授权

US07523032B2 Speech coding method, device, coding module, system and software program product for pre-processing the phase structure of a to be encoded speech signal to match the phase structure of the decoded signal 失效
标题翻译：用于对要编码的语音信号的相位结构进行预处理以匹配解码信号的相位结构的语音编码方法，装置，编码模块，系统和软件程序产品
公开(公告)号：US07523032B2
公开(公告)日：2009-04-21
申请号：US10742645
申请日：2003-12-19
申请人： Ari Heikkinen , Sakari Himanen , Anssi Rämö
发明人： Ari Heikkinen , Sakari Himanen , Anssi Rämö
IPC分类号： G10L11/04 , G10L19/10 , G10L19/04
CPC分类号： G10L19/265 , G10L19/08 , G10L19/16
摘要： The invention relates to a method for use in parametric speech coding. In order to enable an improved parametric coding of speech signals, the method comprises a first step of pre-processing a to be encoded speech based signal such that a phase structure of the to be encoded speech based signal is approached to a phase structure which is obtained when the to be encoded speech based signal is parametrically encoded and decoded again. Only in a second step, a parametric encoding is applied to this pre-processed to be encoded speech based signal. The invention relates equally to a corresponding device, to a corresponding coding module, to a corresponding system and to a corresponding software program product.
摘要翻译：本发明涉及一种用于参数语音编码的方法。为了能够实现语音信号的改进的参数编码，该方法包括对要编码的基于语音的信号进行预处理的第一步骤，使得要编码的基于语音的信号的相位结构接近于相位结构，该相位结构是当要编码的基于语音的信号被再次参数编码和解码时获得。仅在第二步骤中，将参数编码应用于该预处理为被编码的基于语音的信号。本发明同样涉及对应的设备，相应的编码模块，相应的系统和相应的软件程序产品。

5. 发明授权

US06915257B2 Method and apparatus for speech coding with voiced/unvoiced determination 失效
标题翻译：用语音/清音确定语音编码的方法和装置
公开(公告)号：US06915257B2
公开(公告)日：2005-07-05
申请号：US09740826
申请日：2000-12-21
申请人： Ari Heikkinen , Samuli Pietila , Vesa Ruoppila
发明人： Ari Heikkinen , Samuli Pietila , Vesa Ruoppila
IPC分类号： G10L25/93 , G10L11/06 , G10L11/04
CPC分类号： G10L25/93
摘要： This invention presents a voicing determination algorithm for classification of a speech signal segment as voiced or unvoiced. The algorithm is based on a normalized autocorrelation where the length of the window is proportional to the pitch period. The speech segment to be classified is further divided into a number of sub-segments, and the normalized autocorrelation is calculated for each sub-segment if a certain number of the normalized autocorrelation values is above a predetermined threshold, the speech segment is classified as voiced. To improve the performance of the voicing determination algorithm in unvoiced to voiced transients, the normalized autocorrelations of the last sub-segments are emphasized. The performance of the voicing decision algorithm can be enhanced by utilizing also the possible lookahead information.
摘要翻译：本发明提出了一种用于将语音信号段分类为有声或无声的语音确定算法。该算法基于归一化的自相关，其中窗口的长度与音调周期成比例。要分类的语音段被进一步划分为多个子段，并且如果一定数量的归一化自相关值高于预定阈值，则针对每个子段计算归一化的自相关，该语音段被分类为有声。为了提高无声至浊音瞬态中的发音确定算法的性能，强调了最后一个子段的归一化自相关。可以通过利用可能的前瞻信息来增强语音决策算法的性能。

6. 发明申请

US20050137858A1 Speech coding 失效
标题翻译：语音编码
公开(公告)号：US20050137858A1
公开(公告)日：2005-06-23
申请号：US10742645
申请日：2003-12-19
申请人： Ari Heikkinen , Sakari Himanen , Anssi Ramo
发明人： Ari Heikkinen , Sakari Himanen , Anssi Ramo
IPC分类号： G10L19/06 , G10L19/08 , G10L19/14
CPC分类号： G10L19/265 , G10L19/08 , G10L19/16
摘要： The invention relates to a method for use in parametric speech coding. In order to enable an improved parametric coding of speech signals, the method comprises a first step of pre-processing a to be encoded speech based signal such that a phase structure of the to be encoded speech based signal is approached to a phase structure which is obtained when the to be encoded speech based signal is parametrically encoded and decoded again. Only in a second step, a parametric encoding is applied to this pre-processed to be encoded speech based signal. The invention relates equally to a corresponding device, to a corresponding coding module, to a corresponding system and to a corresponding software program product.
摘要翻译：本发明涉及一种用于参数语音编码的方法。为了能够实现语音信号的改进的参数编码，该方法包括对要编码的基于语音的信号进行预处理的第一步骤，使得要编码的基于语音的信号的相位结构接近于相位结构，该相位结构是当要编码的基于语音的信号被再次参数编码和解码时获得。仅在第二步骤中，将参数编码应用于该预处理为被编码的基于语音的信号。本发明同样涉及对应的设备，相应的编码模块，相应的系统和相应的软件程序产品。

7. 发明申请

US20050091044A1 Method and system for pitch contour quantization in audio coding 审中-公开
标题翻译：音频编码中音调轮廓量化的方法和系统
公开(公告)号：US20050091044A1
公开(公告)日：2005-04-28
申请号：US10692291
申请日：2003-10-23
申请人： Anssi Ramo , Jani Nurminen , Sakari Himanen , Ari Heikkinen
发明人： Anssi Ramo , Jani Nurminen , Sakari Himanen , Ari Heikkinen
IPC分类号： G10L11/04 , G10L19/02 , H03M20060101
CPC分类号： G10L19/032 , G10L19/09
摘要： A method and device for improving coding efficiency in audio coding. From the pitch values of a pitch contour of an audio signal, a plurality of simplified pitch contour segments are generated to approximate the pitch contour, based on one or more pre-selected criteria. The contour segments can be linear or non-linear with each contour segment represented by a first end point and a second end point. If the contour segments are linear, then only the information regarding the end points, instead of the pitch values, are provided to a decoder for reconstructing the audio signal. The contour segment can have a fixed maximum length or a variable length, but the deviation between a contour segment and the pitch values in that segment is limited by a maximum value.
摘要翻译：一种提高音频编码效率的方法和装置。根据音频信号的音调轮廓的音调值，基于一个或多个预先选择的标准，生成多个简化俯仰轮廓线段以近似俯仰轮廓。轮廓段可以是由第一终点和第二终点表示的每个轮廓段线性或非线性的。如果轮廓段是线性的，则仅将关于终点而不是音调值的信息提供给用于重建音频信号的解码器。轮廓段可以具有固定的最大长度或可变长度，但轮廓段与该段中的俯仰值之间的偏差受到最大值的限制。

8. 发明申请

US20050091041A1 Method and system for speech coding 审中-公开
标题翻译：语音编码方法和系统
公开(公告)号：US20050091041A1
公开(公告)日：2005-04-28
申请号：US10692290
申请日：2003-10-23
申请人： Anssi Ramo , Jani Nurminen , Sakari Himanen , Ari Heikkinen
发明人： Anssi Ramo , Jani Nurminen , Sakari Himanen , Ari Heikkinen
IPC分类号： G10L20060101 , G10L11/06 , G10L19/02 , G10L19/04 , G10L19/14 , G10L21/04 , H04B1/06 , H04M11/00
CPC分类号： G10L19/24
摘要： A method and device for use in conjunction with an encoder for encoding an audio signal into a plurality of parameters. Based on the behavior of the parameters, such as pitch, voicing, energy and spectral amplitude information of the audio signal, the audio signal can be segmented, so that the parameter update rate can be optimized. The parameters of the segmented audio signal are recorded in a storage medium or transmitted to a decoder so as to allow the decoder to reconstruct the audio signal based on the parameters indicative of the segment audio signals. For example, based on the pitch characteristic, the pitch contour can be approximated by a plurality of contour segments. An adaptive downsampling method is used to update the parameters based on the contour segments so as to reduce the update rate. At the decoder, the parameters are updated at the original rate.
摘要翻译：一种与用于将音频信号编码为多个参数的编码器结合使用的方法和装置。基于音频信号的音调，发音，能量和频谱幅度信息等参数的行为，可以对音频信号进行分段，从而可以优化参数更新速率。分段音频信号的参数被记录在存储介质中或被发送到解码器，以便允许解码器基于指示段音频信号的参数重建音频信号。例如，基于俯仰特性，俯仰轮廓可以由多个轮廓段近似。使用自适应下采样方法根据轮廓段更新参数，以便降低更新速率。在解码器处，参数以原始速率更新。

9. 发明授权

US06801887B1 Speech coding exploiting the power ratio of different speech signal components 失效
标题翻译：语音编码利用不同语音信号分量的功率比
公开(公告)号：US06801887B1
公开(公告)日：2004-10-05
申请号：US09666971
申请日：2000-09-20
申请人： Ari Heikkinen , Mikko Tammi , Jani Nurminen
发明人： Ari Heikkinen , Mikko Tammi , Jani Nurminen
IPC分类号： G10L1914
CPC分类号： G10L19/097 , G10L19/24
摘要： A method and system for waveform interpolation speech coding. The method comprises the steps of decomposing the speech signal into a slowly evolving waveform component and a rapidly evolving waveform component in the encoder and determining the power ratio of these surface components so that the power ratio can be used to determine the bit allocation when the surface components are quantized. The power ratio can also be used to modify the phases of the slowly evolving waveform component when the surface components are reconstructed in the decoder in order to improve the speech quality.
摘要翻译：一种用于波形插值语音编码的方法和系统。该方法包括以下步骤：将语音信号分解成编码器中缓慢演变的波形分量和快速演变的波形分量，并确定这些表面分量的功率比，使得当表面的比特分配时可以使用功率比来确定比特分配组分被量化。当在解码器中重构表面分量以便改善语音质量时，功率比也可用于修改缓慢演变的波形分量的相位。

10. 发明授权

US06584437B2 Method and apparatus for coding successive pitch periods in speech signal 有权
公开(公告)号：US06584437B2
公开(公告)日：2003-06-24
申请号：US09878762
申请日：2001-06-11
申请人： Ari Heikkinen , Vesa T. Ruoppila , Samuli Pietilä
发明人： Ari Heikkinen , Vesa T. Ruoppila , Samuli Pietilä
IPC分类号： G10L1104
CPC分类号： G10L19/08
摘要： A method and apparatus for coding successive pitch periods of a speech signal. Based on a priori knowledge of statistical properties of successive speech periods, a shaped lattice structure is designed to cover the most probable points in the pitch space. The codebook index search starts with finding an open-loop estimate in the pitch space considering all dimensions and refining the open-loop estimate in a closed-loop search separately in each dimension based on the shaped lattice structure. The closed-loop search for the first subframe is for obtaining an absolute pitch period or a delta pitch while the closed-loop search for each of the other subframes is for obtaining a delta pitch for the respective subframe.

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