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1. 发明申请

US20080120098A1 Complexity Adjustment for a Signal Encoder 审中-公开
标题翻译：信号编码器的复杂性调整
公开(公告)号：US20080120098A1
公开(公告)日：2008-05-22
申请号：US11562067
申请日：2006-11-21
申请人： Jari M. Makinen , Juha Marila , Hannu J. Mikkola , Janne Vainio , Tuomas Vaittinen , Sakari Himanen , Kai K. Samposalo
发明人： Jari M. Makinen , Juha Marila , Hannu J. Mikkola , Janne Vainio , Tuomas Vaittinen , Sakari Himanen , Kai K. Samposalo
IPC分类号： G10L19/00
CPC分类号： G10L19/22
摘要： The present invention provides, methods, computer-readable media, and apparatuses for tuning and adjusting the computational complexity of algorithm that is executed by a signal encoder. The signal encoder may comprise a speech encoder. When a resource shortage on a computer platform is detected, a degree of the resource shortage and a corresponding complexity adjustment for a speech encoder are determined. The speech encoder is then tuned to adjust the computational complexity of an executed speech processing algorithm. The resource shortage may correspond to a computational capability, audio buffer memory, or battery of a mobile device. A speech process being executed by the mobile device is tuned to adjust the computational demands in accordance with a complexity adjustment. A number of iteration rounds may be adjusted while the speech encoder is executing a speech processing algorithm. The iterations may correspond to an algebraic codebook search.
摘要翻译：本发明提供了用于调整和调整由信号编码器执行的算法的计算复杂度的方法，计算机可读介质和装置。信号编码器可以包括语音编码器。当检测到计算机平台上的资源短缺时，确定了语音编码器的资源短缺程度和对应的复杂度调整。然后调谐语音编码器以调整执行的语音处理算法的计算复杂度。资源短缺可能对应于移动设备的计算能力，音频缓冲存储器或电池。调整由移动设备执行的语音过程以根据复杂性调整来调整计算需求。当语音编码器执行语音处理算法时，可以调整多个迭代轮。迭代可以对应于代数码本搜索。

2. 发明授权

US07495585B2 Method for inputting characters in electronic device 有权
标题翻译：在电子设备中输入字符的方法
公开(公告)号：US07495585B2
公开(公告)日：2009-02-24
申请号：US11433090
申请日：2006-05-12
申请人： Janne Vainio , Hannu J. Mikkola , Hannu Korhonen , Sakari Himanen , Toni P. Nieminen , Tuomas Vaittinen , Juha Marila
发明人： Janne Vainio , Hannu J. Mikkola , Hannu Korhonen , Sakari Himanen , Toni P. Nieminen , Tuomas Vaittinen , Juha Marila
IPC分类号： H03M1/22
CPC分类号： G06F3/0233 , G06F3/167 , H04M1/23 , H04M2250/70
摘要： According to an aspect of the invention, an enhanced audible feedback solution has been invented for electronic devices using an input device facilitating navigation though a plurality of available user interface input options and confirmation of a selected input option. The electronic device is arranged to define, as a response to detecting a selection of a character on the basis of a detection of a first input to an input device of the electronic device, an audio segment specific to the character. The electronic device is arranged to output the defined audio segment via the audio output means prior to a confirmation by a second input to the input device, the second input being associated with a function adding the character as part of a character sequence entered by the user.
摘要翻译：根据本发明的一个方面，已经针对电子设备发明了一种增强的可听反馈解决方案，所述输入设备通过多个可用的用户接口输入选项和所选择的输入选项的确认来促进导航。电子设备被配置为根据对电子设备的输入设备的第一输入的检测来检测字符的选择的响应来定义特定于该字符的音频段。电子设备被布置成在通过输入设备的第二输入的确认之前经由音频输出装置输出定义的音频片段，第二输入与添加作为用户输入的字符序列的一部分的功能相关联的功能。

3. 发明申请

US20060267931A1 Method for inputting characters in electronic device 有权
公开(公告)号：US20060267931A1
公开(公告)日：2006-11-30
申请号：US11433090
申请日：2006-05-12
申请人： Janne Vainio , Hannu Mikkola , Hannu Korhonen , Sakari Himanen , Toni Nieminen , Tuomas Vaittinen , Juha Marila
发明人： Janne Vainio , Hannu Mikkola , Hannu Korhonen , Sakari Himanen , Toni Nieminen , Tuomas Vaittinen , Juha Marila
IPC分类号： G09G5/00
CPC分类号： G06F3/0233 , G06F3/167 , H04M1/23 , H04M2250/70
摘要： According to an aspect of the invention, an enhanced audible feedback solution has been invented for electronic devices using an input device facilitating navigation though a plurality of available user interface input options and confirmation of a selected input option. The electronic device is arranged to define, as a response to detecting a selection of a character on the basis of a detection of a first input to an input device of the electronic device, an audio segment specific to the character. The electronic device is arranged to output the defined audio segment via the audio output means prior to a confirmation by a second input to the input device, the second input being associated with a function adding the character as part of a character sequence entered by the user.

4. 发明申请

US20070011009A1 Supporting a concatenative text-to-speech synthesis 审中-公开
标题翻译：支持连贯的文本到语音合成
公开(公告)号：US20070011009A1
公开(公告)日：2007-01-11
申请号：US11177250
申请日：2005-07-08
申请人： Jani Nurminen , Sakari Himanen , Anssi Ramo , Janne Vainio
发明人： Jani Nurminen , Sakari Himanen , Anssi Ramo , Janne Vainio
IPC分类号： G10L13/08
CPC分类号： G10L13/06
摘要： The invention relates to a support of a concatenative TTS synthesis. In order to generate a speech database as a basis for the TTS synthesis, first, a speech processing including a segmental parametric speech encoding of speech data based on a parametric modeling of speech is performed, which results in compressed parameterized speech segments. Then, the compressed parameterized speech segments are assembled in a speech database. In order to synthesize output speech, compressed parameterized speech segments are selected from the speech database based on an available text and decompressed to regain parameterized speech segments. The parameterized speech segments are then concatenated in a parameter domain. The output speech is synthesized based on these concatenated parametric speech segments.
摘要翻译：本发明涉及一种级联TTS合成的支持。为了生成语音数据库作为TTS综合的基础，首先，执行包括基于语音的参数建模的语音数据的分段参数语音编码的语音处理，这导致压缩的参数化语音段。然后，压缩的参数化语音段被组合在语音数据库中。为了合成输出语音，基于可用文本从语音数据库中选择压缩的参数化语音段，并且解压缩以重新获得参数化语音段。参数化语音段然后在参数域中连接。基于这些连接的参数语音段来合成输出语音。

5. 发明授权

US08489392B2 System and method for modeling speech spectra 有权
标题翻译：语音谱建模系统和方法
公开(公告)号：US08489392B2
公开(公告)日：2013-07-16
申请号：US11855108
申请日：2007-09-13
申请人： Jani Nurminen , Sakari Himanen
发明人： Jani Nurminen , Sakari Himanen
IPC分类号： G10L11/06
CPC分类号： G10L25/93 , G10L19/0204 , G10L2025/935
摘要： A system and method for modeling speech in such a way that both voiced and unvoiced contributions can co-exist at certain frequencies. In various embodiments, three spectral bands (or bands of up to three different types) are used. In one embodiment, the lowest band or group of bands is completely voiced, the middle band or group of bands contains both voiced and unvoiced contributions, and the highest band or group of bands is completely unvoiced. The embodiments of the present invention may be used for speech coding and other speech processing applications.
摘要翻译：一种用于对语音进行建模的系统和方法，使得有声和无声的贡献可以在某些频率下共存。在各种实施例中，使用三个光谱带（或多达三种不同类型的频带）。在一个实施例中，最低频带或频带组完全浊音，中间频带或频带组包含有声和无声贡献，并且最高频带或组的频带完全无声。本发明的实施例可以用于语音编码和其他语音处理应用。

6. 发明授权

US08380496B2 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：US08380496B2
公开(公告)日：2013-02-19
申请号：US12150307
申请日：2008-04-25
申请人： Anssi Rämö , Jani Nurminen , Sakari Himanen , Ari Heikkinen
发明人： Anssi Rämö , Jani Nurminen , Sakari Himanen , Ari Heikkinen
IPC分类号： G10L11/04 , G10L19/00
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。

7. 发明申请

US20060080090A1 Reusing codebooks in parameter quantization 审中-公开
标题翻译：在参数量化中重用码本
公开(公告)号：US20060080090A1
公开(公告)日：2006-04-13
申请号：US10961471
申请日：2004-10-07
申请人： Anssi Ramo , Sakari Himanen , Jani Nurminen
发明人： Anssi Ramo , Sakari Himanen , Jani Nurminen
IPC分类号： G10L19/12
CPC分类号： G10L19/07
摘要： The present invention provides a new methodology for reusing codebooks for a multistage vector quantization of parameter quantizers of signals. Prior art multistage vector quantization is done in such a way that each stage has different optimized codebooks. The prior art codebooks, thus, use quite a lot of a memory storage space. Using the same codebook stages several times, according to the present invention, reduces the memory usage and a codebook structure maintains good quality by using optimized codebooks for the most important (first) stages in the quantization. The number of codebooks is reduced by reusing the same codebooks in the refining stages. Additionally, according to the present invention, using many predictors is space-wise efficient since they need only a few of coefficients instead of larger codebooks.
摘要翻译：本发明提供了一种用于重新使用信号参数量化器的多级矢量量化码本的新方法。现有技术的多级矢量量化是以每一级具有不同优化码本的方式完成的。因此，现有技术的码本使用相当多的存储器存储空间。使用相同的码本阶段，根据本发明，通过使用量化中最重要的（第一）级的优化码本来减少存储器使用并且码本结构保持良好的质量。通过在精炼阶段重复使用相同的码本来减少码本的数量。此外，根据本发明，使用许多预测器是空间有效的，因为它们仅需要少数系数而不是较大的码本。

8. 发明授权

US08086057B2 Dynamic quantizer structures for efficient compression 有权
标题翻译：用于高效压缩的动态量化器结构
公开(公告)号：US08086057B2
公开(公告)日：2011-12-27
申请号：US11855778
申请日：2007-09-14
申请人： Jani Nurminen , Sakari Himanen
发明人： Jani Nurminen , Sakari Himanen
IPC分类号： G06K9/00
CPC分类号： H04N19/126 , G10L19/032 , H04N19/46
摘要： A method and system are introduced that provide dynamic quantizer structures which are configurable during run time. A quantizer configuration and data are stored in a binary format. The dynamic quantizer data is represented as a bitstream, and the bitstream in turn is used as additional input during initialization (or re-initialization/re-configuration) of a speech coder. A configuration header fully specifies the structure and configuration of the dynamic quantizer for each quantized parameter, and the dynamic quantizer data and configurations are fully and dynamically allocated into the speech coder memory. This enables easy re-configuration of a codec associated with the quantizer structures for different scenarios. The use of dynamic quantizer structures in turn enhances compression efficiency of an input signal. The dynamic quantizer structures can also be applied to other compression applications that allow lossy compression.
摘要翻译：引入了一种提供在运行时可配置的动态量化器结构的方法和系统。量化器配置和数据以二进制格式存储。动态量化器数据被表示为比特流，并且在语音编码器的初始化（或重新初始化/重新配置）期间，比特流又被用作附加输入。配置头完全指定每个量化参数的动态量化器的结构和配置，动态量化器数据和配置被完全和动态地分配到语音编码器存储器中。这使得能够容易地重新配置与用于不同场景的量化器结构相关联的编解码器。动态量化器结构的使用又提高了输入信号的压缩效率。动态量化器结构也可以应用于允许有损压缩的其他压缩应用。

9. 发明申请

US20090094264A1 Method, Apparatus and Computer Program Product for Providing Improved Data Compression 有权
标题翻译：提供改进数据压缩的方法，设备和计算机程序产品
公开(公告)号：US20090094264A1
公开(公告)日：2009-04-09
申请号：US11867212
申请日：2007-10-04
申请人： Jani K. Nurminen , Sakari Himanen
发明人： Jani K. Nurminen , Sakari Himanen
IPC分类号： G06F17/30
CPC分类号： H04N19/124 , G06T9/00 , G10L19/032 , G10L19/24 , H04N19/147 , H04N19/196
摘要： An apparatus for providing improved data compression may include an encoder comprising a quantizer for encoding input data and a side model. The quantizer may be trained with respect to high priority data among the input data and may be configured to partially encode the input data by encoding the high priority data. The side model may be trained jointly with the training of the quantizer and is configured to model low priority data among the input data.
摘要翻译：用于提供改进的数据压缩的装置可以包括编码器，其包括用于对输入数据进行编码的量化器和侧模型。量化器可以相对于输入数据中的高优先级数据进行训练，并且可以被配置为通过对高优先级数据进行编码来对输入数据进行部分编码。侧模型可以与量化器的训练联合进行训练，并且被配置为对输入数据中的低优先级数据进行建模。

10. 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。

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